0) Quick choices & prerequisites (decide these first)

Model family: wav2vec 2.0 (self-supervised, fine-tuning works very well).

(arXiv)

Datasets: Mozilla Common Voice (large, many languages) and LibriSpeech (clean benchmark).
([Link])
Framework: PyTorch + Hugging Face Transformers + datasets + torchaudio. Hugging Face has a
concrete fine-tuning tutorial you can follow. (Hugging Face)
Compute: 1× 12–24 GB GPU (RTX 3080/4090 or A100 for faster training). Smaller GPUs will require
gradient accumulation and smaller batches.
Language: Start with English — multilingual adds complexity.

1) Project folder structure

stt_project/
├─ data/
│ ├─ common_voice/ # raw downloads
│ └─ librispeech/
├─ datasets/ # processed manifests (wav, transcript)
├─ models/
│ ├─ checkpoints/
├─ src/
│ ├─ [Link]
│ ├─ data_collator.py
│ ├─ train_wav2vec.py
│ ├─ [Link]
│ └─ serve_api.py
├─ [Link]
└─ notebooks/

2) Get & prepare datasets

 Download LibriSpeech (OpenSLR) and Mozilla Common Voice. Use
torchaudio / huggingface datasets to simplify loading. ([Link])

Standardize audio format: 16 kHz, mono, WAV PCM16. Many pretrained

wav2vec2 checkpoints expect 16kHz.
Example (ffmpeg):

ffmpeg -i input.mp3 -ac 1 -ar 16000 [Link]

Create manifests: CSV/JSON with columns path, sentence, duration, speaker_id.

Hugging Face datasets can store these as dataset objects.

Clean transcripts:

Lowercase, remove punctuation (depending on tokenizer), normalize

numbers if desired.

Strip long silence-only files and very short clips (<0.5s).

Split into train / validation / test (if using dataset that does not provide splits, keep 5–10% validation).

3) Choose approach: fine-tune pretrained

wav2vec2 (recommended)
Why: wav2vec 2.0 learns speech representation from raw audio and fine-tunes well
with much less labeled data. Use an existing checkpoint (facebook/wav2vec2-large-
960h or smaller) and fine-tune on your dataset. (Hugging Face)

High-level steps:
 Tokenizer / vocabulary: build char-level tokenizer (a–z, space, apostrophe) or use byte-
level; Hugging Face examples use Wav2Vec2CTCTokenizer.

Feature extractor: Wav2Vec2FeatureExtractor (handles sampling/normalization).

Model: Wav2Vec2ForCTC from Transformers, with final linear layer size = vocab size.

Data collator: pad audio to batch, compute input values + labels, mask labels for
padding.

Useful tutorial: Hugging Face "Fine-Tune Wav2Vec2 for English ASR". (Hugging Face)

4) Example code outline (key parts)

Install deps:

pip install torch torchaudio transformers datasets jiwer soundfile

Data loading + preprocessing (simplified):

from datasets import load_dataset, load_metric

from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
# load Common Voice via huggingface dataset (example)
ds = load_dataset("mozilla-foundation/common_voice_11_0", "en", split='train+validation')
# Resample & keep fields: [Link], text
def prepare_batch(batch):
audio = batch["audio"]
batch["input_values"] = processor(audio["array"],
sampling_rate=audio["sampling_rate"]).input_values[0]
batch["labels"] = [Link](batch["sentence"]).input_ids
return batch
Training scaffolding: use Trainer OR custom training loop. Hugging Face Trainer
simplifies things.

Key hyperparams (baseline):

lr: 1e-4 (with warmup)

batch_size: 8–32 (depending on GPU)

epochs: 10–30 (watch val WER)

gradient_accumulation_steps if small GPU

eval steps: every 500–1000 steps

5) Evaluation: WER & CER

Use jiwer or Hugging Face evaluate to compute Word Error Rate (WER) and Character
Error Rate (CER).

Keep a held-out test set (LibriSpeech test-clean / test-other) for benchmark.

Example:
from jiwer import wer
preds = ["hello world", "test"]
refs = ["hello world", "test"]
print(wer(refs, preds))
Aim: WER < 10–15% on clean data for a fine-tuned small model; fine models on lots
of data achieve single-digit WER.

6) Improve performance (practical tips)

Data augmentation: Add noise, speed perturbation, reverberation (Kaldi style) —

improves robustness.

Domain fine-tuning: fine-tune on domain-specific audio (accent, microphone type).

Language Model (LM): Use a shallow n-gram LM (KenLM) for decoding to reduce WER
(optional). Standalone wav2vec2 does OK without LM but LM helps in noisy/cross-
domain cases.

SpecAugment: time & frequency masking for robustness.

Curriculum: start training on clean data then include noisy data.

Gradient checkpointing and mixed precision (fp16) to reduce memory.

7) Exporting the trained model for inference

Save the Hugging Face processor and Wav2Vec2ForCTC checkpoint:
model.save_pretrained("models/my_wav2vec2")
processor.save_pretrained("models/my_wav2vec2")
For fast inference, convert to TorchScript / ONNX (optional) for C++ or lower-latency serving.
8) Inference server + API (to connect with
Unreal)
Build a Flask/FastAPI endpoint that accepts audio (wav) and returns the transcript
(and optional confidences/timestamps):

Example using FastAPI:

from fastapi import FastAPI, File, UploadFile

import soundfile as sf
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
app = FastAPI()
processor = Wav2Vec2Processor.from_pretrained("models/my_wav2vec2")
model = Wav2Vec2ForCTC.from_pretrained("models/my_wav2vec2").to("cuda")
@[Link]("/transcribe")
async def transcribe(file: UploadFile = File(...)):
data, sr = [Link]([Link])
input_values = processor(data, sampling_rate=sr, return_tensors="pt").input_values.to("cuda")
with torch.no_grad():
logits = model(input_values).logits
pred_ids = [Link](logits, dim=-1)
transcription = processor.batch_decode(pred_ids)[0]
return {"text": transcription}

Unreal can POST user audio to [Link] and receive text.

9) Latency & resource considerations

On GPU, real-time or near real-time is achievable (depends on model size). On CPU,

large models may be slow. Use smaller base models or quantize for CPU deployment.
Use streaming inference (chunked audio + stateful decoding) for live microphone input. Hugging Face
has streaming recipes; for production you may implement VAD + short chunks.

10) Alternatives & fallbacks

If training/fine-tuning is too slow for the project timeline, use Vosk or prebuilt offline
models to prototype and switch to your fine-tuned wav2vec later. Vosk is easy to run
offline and supports many languages. ([Link])

Coqui STT and other open toolkits are options if you prefer TensorFlow setups.

11) Example timeline (6–8 weeks realistic for MVP)

Week 1: Download & preprocess datasets (Common Voice subset or LibriSpeech). Create
manifests.

Week 2: Set up training code + small overfit test (single file) to verify pipeline.

Week 3–4: Fine-tune pretrained wav2vec2 on your dataset; iterate hyperparams.

Week 5: Evaluate (WER/CER), add augmentation, LM decode experiments.

Week 6: Build inference API + test with Unreal integration (send recorded wavs).

Week 7–8: Polish, optimize latency, finalize report & demo.

12) Common pitfalls & debugging checklist

 Mismatch sample rate → get weird transcriptions. Ensure 16 kHz everywhere.

Label/text cleanup issues (extra punctuation) → tokenizer mismatch.

Batch padding mistakes → label shifting and loss spikes. Use HF data collator examples.

Overfitting small dataset → use augmentation and validation split.

GPU OOM → reduce batch size, enable fp16 or gradient accumulation.

13) Useful links & references

Wav2Vec 2.0 paper & code (Meta). (arXiv)

Hugging Face fine-tune tutorial (practical notebook). (Hugging Face)

Mozilla Common Voice dataset (download & releases). ([Link])

LibriSpeech corpus (OpenSLR). ([Link])

Vosk (offline alternative). ([Link])

If you want, I can next provide any of the following immediately (pick one — I’ll produce
it now):
 A ready-to-run training script (Hugging Face Trainer + data collator) tuned for Common
Voice.

A FastAPI inference server (complete file) that returns transcripts and confidence.

A step-by-step notebook that shows dataset loading → preprocessing → tiny fine-tune

(so you can run quickly on a small subset).

A Unreal blueprint + Python API example that sends microphone audio and plays
back transcribed text.

Which one do you want me to produce right away?