suleiman2003/mms-tts-hau-2speaker

Multi-Speaker MMS-TTS Hausa Model (2 Speakers)

This model is a structurally converted version of facebook/mms-tts-hau adapted for 2-speaker text-to-speech synthesis.

⚠️ IMPORTANT: This model requires fine-tuning before use. Speaker embeddings are randomly initialized.

Model Description

• Architecture: VITS (Variational Inference with adversarial learning for end-to-end Text-to-Speech)
• Language: Hausa (ISO 639-3: hau)
• Number of Speakers: 2
• Speaker Embedding Dimension: 256
• Base Model: facebook/mms-tts-hau
• Status: ⚠️ Requires Fine-tuning

Architecture Details

VITS Components

The VITS architecture consists of:

1. Text Encoder (Preserved from original)

   • Transformer-based text encoder
   • Converts text to linguistic features
   • Language-specific, not speaker-dependent

2. Posterior Encoder (Reinitialized)

   • Encodes speaker characteristics from reference audio
   • Learns speaker-specific acoustic features
   • Must be trained on your speaker data

3. Flow-based Module (Reinitialized)

   • Normalizing flows for feature transformation
   • Learns speaker-dependent acoustic mappings
   • Consists of coupling layers

4. Decoder/Generator (Reinitialized)

   • Generates final waveform
   • Speaker-specific synthesis
   • Based on HiFi-GAN vocoder

5. Speaker Embedding Layer (Newly Added)

   • 2 speaker embeddings
   • 256-dimensional vectors
   • Randomly initialized

What Was Changed

✓ Preserved:
  - Text encoder (language understanding)
  - Model tokenizer
  - Base architecture

✗ Reinitialized:
  - Posterior encoder (speaker characteristics)
  - Flow-based transformations
  - Decoder/generator
  - Duration predictor

+ Added:
  - Speaker embedding layer (2x256)

Fine-tuning Required

This model will not work without fine-tuning. You need:

1. Dataset: Multi-speaker Hausa TTS dataset with:

   • Audio files (16kHz recommended)
   • Transcriptions in Hausa
   • Speaker labels (0 and 1)
   • At least 50-150 utterances per speaker

2. Fine-tuning: Use the finetune-hf-vits repository

Fine-tuning Steps

# Clone fine-tuning repository
git clone https://github.com/ylacombe/finetune-hf-vits.git
cd finetune-hf-vits

# Install requirements
pip install -r requirements.txt

# Build monotonic alignment search
cd monotonic_align
python setup.py build_ext --inplace
cd ..

# Run fine-tuning (adjust parameters as needed)
accelerate launch run_vits_finetuning.py \
  --model_name_or_path "suleiman2003/mms-tts-hau-2speaker" \
  --dataset_name "your-dataset" \
  --output_dir "./output" \
  --num_train_epochs 100 \
  --learning_rate 2e-4 \
  --warmup_ratio 0.0 \
  --per_device_train_batch_size 8 \
  --speaker_id_column_name "speaker_id"

Usage (After Fine-tuning)

from transformers import VitsModel, VitsTokenizer, set_seed
import torch
import scipy

# Load fine-tuned model
model = VitsModel.from_pretrained("suleiman2003/mms-tts-hau-2speaker")
tokenizer = VitsTokenizer.from_pretrained("suleiman2003/mms-tts-hau-2speaker")

# Prepare input
text = "Sannu, yaya kake?"  # "Hello, how are you?" in Hausa
inputs = tokenizer(text, return_tensors="pt")

# Generate speech for speaker 0
set_seed(42)
with torch.no_grad():
    outputs = model(**inputs, speaker_id=0)

# Save audio
audio = outputs.waveform[0].cpu().numpy()
scipy.io.wavfile.write("output_speaker0.wav", rate=16000, data=audio)

# Generate speech for speaker 1
with torch.no_grad():
    outputs = model(**inputs, speaker_id=1)
audio = outputs.waveform[0].cpu().numpy()
scipy.io.wavfile.write("output_speaker1.wav", rate=16000, data=audio)

Dataset Preparation

Your dataset should follow this structure:

# Example dataset format
{
    "audio": [<audio_path_1>, <audio_path_2>, ...],
    "text": ["Yaya kake?", "Ina kwana?", ...],
    "speaker_id": [0, 1, 0, 1, ...]  # Speaker labels
}

Recommended Datasets

• Create your own Hausa multi-speaker dataset
• Use existing Hausa corpora and add speaker annotations
• Minimum 100 utterances per speaker (more is better)

Technical Specifications

• Sampling Rate: 16,000 Hz
• Model Parameters: ~83M
• Framework: PyTorch + Transformers
• Training Method: GAN-based (Generator + Discriminator)

Limitations

• Requires fine-tuning: Cannot generate speech without training
• Speaker embeddings: Currently random, need to learn from data
• Language-specific: Optimized for Hausa only
• License: Inherits CC-BY-NC-4.0 (non-commercial) from base model

Citation

If you use this model, please cite:

@article{pratap2023mms,
  title={Scaling Speech Technology to 1,000+ Languages},
  author={Pratap, Vineel and Tjandra, Andros and Shi, Bowen and others},
  journal={arXiv preprint arXiv:2305.13516},
  year={2023}
}

@inproceedings{kim2021vits,
  title={Conditional variational autoencoder with adversarial learning for end-to-end text-to-speech},
  author={Kim, Jaehyeon and Kong, Jungil and Son, Juhee},
  booktitle={International Conference on Machine Learning},
  pages={5530--5540},
  year={2021},
  organization={PMLR}
}

Resources

• Base Model: facebook/mms-tts-hau
• Fine-tuning Guide: finetune-hf-vits
• VITS Paper: arXiv:2106.06103
• MMS Paper: arXiv:2305.13516

License

This model inherits the CC-BY-NC-4.0 license from the base MMS model, restricting commercial use.