Oriserve / Whisper-Hindi2Hinglish-Apex

Whisper-Hindi2Hinglish-Apex:

• GITHUB LINK: github link
• SPEECH-TO-TEXT ARENA: Speech-To-Text Arena

Table of Contents:

• Key Features
• Training
  • Data
  • Finetuning
• Usage
• Performance Overview
  • Qualitative Performance Overview
  • Quantitative Performance Overview
• Miscellaneous

Key Features:

1. Hinglish as a language : Native ability to transcribe hindi into hinglish, making it easier to understand and process.
2. Faster Inference : The model achieves SOTA performance while being upto 8x faster.
3. Performance Increase : ~42% average performance increase versus pretrained model across benchmarking datasets.
4. Ranked #1 on Speech-To-Text Arena , beating competitors on transcription preference metric.

Training:

Data:

• Duration : A total of ~700 Hrs of noisy Indian-accented Hindi audios.
• Collection : Curated using a collection of open-source and proprietary dataset.
• Labelling : Automated labelling using SOTA model, with manual human correction.

Finetuning:

• Novel Finetuning Techniques : Advanced fine-tuning techniques were used to specifically target and increase performance on noisy Indian-accented audios.
  • Custom Dynamic Layer Freezing : Identifying most active layers during inference and perfomed targeted training of those layers.
  • Dataset Augmentations : Devised data augmentation techniques to increase model robustness.

Performance Overview

Qualitative Performance Overview

Quantitative Performance Overview

Note :

• To accurately measure Hinglish transcription performance, the original Hindi ground truth for each dataset was first transliterated to Hinglish. The WER scores below were calculated against this transliterated reference text.
• To check our model's real-world performance against other SOTA models please head to our Speech-To-Text Arena arena space.

Usage:

Using Transformers

• To run the model, first install the Transformers library

pip install -U transformers

• The model can be used with the pipeline class to transcribe audios of arbitrary length:

import torch
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
from datasets import load_dataset

# Set device (GPU if available, otherwise CPU) and precision
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32

# Specify the pre-trained model ID
model_id = "Oriserve/Whisper-Hindi2Hinglish-Apex"

# Load the speech-to-text model with specified configurations
model = AutoModelForSpeechSeq2Seq.from_pretrained(
    model_id,
    torch_dtype=torch_dtype,        # Use appropriate precision (float16 for GPU, float32 for CPU)
    low_cpu_mem_usage=True,         # Optimize memory usage during loading
    use_safetensors=True            # Use safetensors format for better security
)
model.to(device)                    # Move model to specified device

# Load the processor for audio preprocessing and tokenization
processor = AutoProcessor.from_pretrained(model_id)

# Create speech recognition pipeline
pipe = pipeline(
    "automatic-speech-recognition",
    model=model,
    tokenizer=processor.tokenizer,
    feature_extractor=processor.feature_extractor,
    torch_dtype=torch_dtype,
    device=device,
    generate_kwargs={
        "task": "transcribe",       # Set task to transcription
        "language": "en"            # Specify English language
    }
)

# Process audio file and print transcription
sample = "sample.wav"               # Input audio file path
result = pipe(sample)               # Run inference
print(result["text"])               # Print transcribed text

Using Flash Attention 2

Flash-Attention 2 can be used to make the transcription fast. If your GPU supports Flash-Attention you can use it by, first installing Flash Attention:

pip install flash-attn --no-build-isolation

• Once installed you can then load the model using the below code:

model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, attn_implementation="flash_attention_2")

Using the OpenAI Whisper module

• First, install the openai-whisper library

pip install -U openai-whisper tqdm

• Convert the huggingface checkpoint to a pytorch model

import torch
from transformers import AutoModelForSpeechSeq2Seq
import re
from tqdm import tqdm
from collections import OrderedDict
import json

# Load parameter name mapping from HF to OpenAI format
with open('convert_hf2openai.json', 'r') as f:
    reverse_translation = json.load(f)

reverse_translation = OrderedDict(reverse_translation)

def save_model(model, save_path):
    def reverse_translate(current_param):
        # Convert parameter names using regex patterns
        for pattern, repl in reverse_translation.items():
            if re.match(pattern, current_param):
                return re.sub(pattern, repl, current_param)

    # Extract model dimensions from config
    config = model.config
    model_dims = {
        "n_mels": config.num_mel_bins,           # Number of mel spectrogram bins
        "n_vocab": config.vocab_size,            # Vocabulary size
        "n_audio_ctx": config.max_source_positions,    # Max audio context length
        "n_audio_state": config.d_model,         # Audio encoder state dimension
        "n_audio_head": config.encoder_attention_heads,  # Audio encoder attention heads
        "n_audio_layer": config.encoder_layers,   # Number of audio encoder layers
        "n_text_ctx": config.max_target_positions,     # Max text context length
        "n_text_state": config.d_model,          # Text decoder state dimension
        "n_text_head": config.decoder_attention_heads,  # Text decoder attention heads
        "n_text_layer": config.decoder_layers,    # Number of text decoder layers
    }

    # Convert model state dict to Whisper format
    original_model_state_dict = model.state_dict()
    new_state_dict = {}

    for key, value in tqdm(original_model_state_dict.items()):
        key = key.replace("model.", "")          # Remove 'model.' prefix
        new_key = reverse_translate(key)         # Convert parameter names
        if new_key is not None:
            new_state_dict[new_key] = value

    # Create final model dictionary
    pytorch_model = {"dims": model_dims, "model_state_dict": new_state_dict}

    # Save converted model
    torch.save(pytorch_model, save_path)

# Load Hugging Face model
model_id = "Oriserve/Whisper-Hindi2Hinglish-Apex"
model = AutoModelForSpeechSeq2Seq.from_pretrained(
    model_id,
    low_cpu_mem_usage=True,        # Optimize memory usage
    use_safetensors=True           # Use safetensors format
)

# Convert and save model
model_save_path = "Whisper-Hindi2Hinglish-Apex.pt"
save_model(model,model_save_path)

• Transcribe

import whisper
# Load converted model with Whisper and transcribe
model = whisper.load_model("Whisper-Hindi2Hinglish-Apex.pt")
result = model.transcribe("sample.wav")
print(result["text"])

Miscellaneous

This model is from a family of transformers-based ASR models trained by Oriserve. To compare this model against other models from the same family or other SOTA models please head to our Speech-To-Text Arena . To learn more about our other models, and other queries regarding AI voice agents you can reach out to us at our email [email protected]