--- language: - hi - bn - te - mr - kn - ta - ml - gu - pa - or - as - en - ur - ks - ne - sd - sa - mai - bho - mag - hne - raj - doi - kok - sat - brx - mni - grt - rwr - bgc - awa - bra - gbm - lmn - bhb - bgq - kfy - xnr - bfy - noe - rjs - mwr - mtr - wbr - hoj - gom - ahr - sgj - kru - unr - spv - kfr - tcy - kfa - sck tags: - speech - asr - automatic-speech-recognition - indian-languages - indic - multilingual - heep license: apache-2.0 library_name: transformers pipeline_tag: automatic-speech-recognition --- # HEEP Indic **High Entropy Exponential Pruning for State-of-the-Art Multilingual ASR** HEEP Indic is a state-of-the-art automatic speech recognition model that demonstrates how strategic entropy-based data curation outperforms brute-force data scaling. With an average word error rate (WER) of **11.9%** on Hindi benchmarks — outperforming Google STT, Azure STT, Nvidia Conformer, and IndicWhisper — it challenges the "more data is better" paradigm by training on carefully selected high-information samples. ## Model Overview HEEP Indic supports transcription across **55 Indic languages**, with consistent performance across various domains such as meetings, earnings calls, broadcast media, and educational content. The model is optimized for high-precision, verbatim transcription capturing spoken content word-for-word with remarkable fidelity. **Core Insight**: Strategic selection of high-entropy samples leads to better ASR models than training on larger but redundant datasets. ## HEEP Methodology HEEP (High Entropy Exponential Pruning) is an entropy-based data curation methodology that prioritizes information density over data quantity. It identifies high-information training samples while progressively filtering redundant data, enabling efficient model training with significantly reduced computational resources. ### Mathematical Foundation #### Sample Score (Equation 1) The information score for each sample combines multiple entropy dimensions: ``` S(x) = α₁·H_acoustic(x) + α₂·H_phonetic(x) + α₃·H_linguistic(x) + α₄·H_contextual(x) + β·MI(x, D) ``` Where: - `H_acoustic(x)`: Spectral/MFCC entropy measuring acoustic diversity - `H_phonetic(x)`: Phoneme distribution entropy capturing phonetic complexity - `H_linguistic(x)`: Vocabulary and syntax entropy measuring linguistic richness - `H_contextual(x)`: Domain and discourse entropy - `MI(x, D)`: Mutual information contribution relative to dataset - `α₁...α₄, β`: Configurable weights (default: 0.25, 0.20, 0.25, 0.15, 0.15) #### Mutual Information (Equation 2) The mutual information between acoustic features and transcription: ``` I(x, y) = Σ_{j,ℓ} p(f_j, y_ℓ) log [p(f_j, y_ℓ) / (p(f_j)·p(y_ℓ))] ``` #### Selection Criterion Samples are selected based on a threshold: ``` D' = {x ∈ D : S(x) > τ} ``` #### Progressive Filtering (Equation 8) The threshold increases exponentially across rounds: ``` τ_{k+1} = τ_k · growth_factor ``` #### Error-Aware Adaptation After each training round, sample scores are adjusted based on model errors: ``` S'(x) = S(x) + λ_err·ErrorRelevance(x, errors_k) + λ_cross·CrossLingualOverlap(x) ``` ### Algorithm Overview ``` Algorithm: HEEP Data Curation with Error-Aware Adaptation Input: Dataset D, initial threshold τ₀, growth factor g Output: Curated dataset D* 1. Initialize scorer with entropy estimators 2. Fit scorer to D (compute normalization stats, fit MI estimator) 3. D* ← D 4. k ← 0 5. While |D*| > min_samples AND k < max_rounds: a. For each x in D*: Compute S(x) = Σᵢ αᵢ·Hᵢ(x) + β·MI(x, D) b. If error_patterns available: Adjust S'(x) = S(x) + λ_err·ErrorRelevance(x) + λ_cross·CrossLingualOverlap(x) c. D* ← {x ∈ D* : S'(x) > τₖ} d. If train_callback: Train model on D* e. If eval_callback: Analyze errors, update error_patterns f. τₖ₊₁ ← τₖ · g g. k ← k + 1 6. Return D* ``` ### Key Benefits - Training on **10-20% of data** while matching or exceeding full-dataset performance - Efficient multilingual model development with cross-lingual transfer - Error-aware adaptive sample selection across training rounds - Significant reduction in computational resources and training time ## Post-Rebuttal Update: Cross-Architecture Validation with HEEP-Indic **Addressing Q1 (Gain Attribution), Q2 (Baselines), and Q3 (Base Model Dependency)** We apologize for the supplementary post after the rebuttal period. These results were finalized shortly after the deadline, and we wanted to ensure complete experimental evidence was available rather than leave placeholders. #### Resources * **Reproducibility (Universal Model):** [https://huggingface.co/bc7ec356/heep-universal](https://huggingface.co/bc7ec356/heep-universal) * **Cross-Architecture Model (Indic):** [https://huggingface.co/bc7ec356/heep-indic](https://huggingface.co/bc7ec356/heep-indic) ### Cross-Architecture Generalization To directly address concerns about generalization beyond Whisper V3 Turbo, we trained **Qwen3-ASR (1.7B)**, an architecturally distinct audio-language model, on HEEP-curated data spanning **46 Indian languages** (~4.78M utterances). The curation pipeline is identical to the one described in the paper with no architecture-specific tuning. ### Hindi Benchmark Comparison (7 Benchmarks) Word error rates (%) on Indic benchmark datasets: | Dataset | Bengali | Bhojpuri | Chhattisgarhi | Gujarati | Hindi | Kannada | Magahi | Maithili | Malayalam | Marathi | Odia | Punjabi | Sanskrit | Tamil | Telugu | Urdu | Avg | |---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---| | Kathbath | 14.6 | – | – | 17.4 | 8.5 | 23 | – | – | 39.3 | 19.2 | 25.4 | 15.8 | 41.4 | 30.3 | 29 | 12.1 | 23 | | Kathbath Hard | 15.7 | – | – | 18.5 | 9 | 25.1 | – | – | 41.2 | 20.4 | 27.7 | 16.6 | 43.6 | 32.6 | 30.3 | 11.9 | 24.4 | | CommonVoice | 21 | – | – | – | 9.96 | – | – | – | 46 | 21.5 | 34.6 | 17.5 | – | 34 | – | 20.6 | 25.7 | | FLEURS | 22.4 | – | – | 23.3 | 11 | 23.1 | – | – | 34.4 | 25.5 | 33.3 | 25 | – | 35.1 | 31.9 | 22.4 | 26.1 | | IndicTTS | 15.8 | – | – | 16.9 | 6.6 | 19.6 | – | – | 26.4 | 14.5 | 14.8 | – | – | 22.6 | 31.3 | – | 18.7 | | Gramvaani | – | – | – | – | 26 | – | – | – | – | – | – | – | – | – | – | – | 26 | | RESPIN | 32.5 | 21.3 | 21.6 | – | 12.1 | 45.6 | 27.7 | 41.1 | – | 32.7 | – | – | – | – | 37.5 | – | 30.2 | | **Average** | **20.4** | **21.3** | **21.6** | **19** | **11.9** | **27.3** | **27.7** | **41.1** | **37.5** | **22.3** | **27.2** | **18.7** | **42.5** | **30.9** | **32** | **16.7** | **24.6** | ### Hindi Benchmark Comparison Comparison of publicly-available models on the Hindi subset of the benchmark: | Model | Kathbath | Kathbath Noisy | CommonVoice | FLEURS | IndicTTS | RESPIN | Gramvaani | Average | |---|---|---|---|---|---|---|---|---| | Google STT | 14.3 | 16.7 | 20.8 | 19.4 | 18.3 | – | 59.9 | 24.9 | | IndicWav2Vec | 12.2 | 16.2 | 20.2 | 18.3 | 15 | – | 42.1 | 20.7 | | Azure STT | 13.6 | 15.1 | 14.6 | 24.3 | 15.2 | – | 42.3 | 20.8 | | Nvidia Conformer-CTC Medium | 14 | 15.6 | 20.4 | 19.4 | 12.3 | – | 41.3 | 20.5 | | Nvidia Conformer-CTC Large | 12.7 | 14.2 | 21.2 | 15.7 | 12.2 | – | 42.6 | 19.8 | | IndicWhisper | 10.3 | 12 | 15 | 11.4 | 7.6 | – | 26.8 | 13.8 | | **HEEP Indic** | **8.53** | **8.97** | **9.96** | **11.04** | **6.59** | **12.05** | **25.98** | **11.9** | **HEEP-Indic achieves 11.9% average Hindi WER vs. 13.8% for IndicWhisper (14% relative improvement).** ### Key Takeaways 1. **Cross-architecture generalization confirmed.** The same HEEP pipeline improves two distinct backbones: Whisper V3 Turbo (0.8B, encoder-decoder) and Qwen3-ASR (1.7B, audio-language model), without modification. 2. **Controlled multilingual evaluation.** Results span 16 languages across Indo-Aryan, Dravidian, and Classical families on standardized benchmarks with consistent evaluation protocols. 3. **Model-independent scoring.** Entropy scoring operates on MFCCs, G2P phonemes, and token distributions, not model internals. The same curated dataset was used for both backbones. 4. **Reproducibility.** Model weights, curation code, and training scripts for both backbones are at the anonymous repository. *We hope Reviewers 2ezj, oXjG, and S4Jd also find this supplementary evidence relevant to their earlier questions on generalization and controlled multilingual evaluation.* ## Model Details - **Architecture**: Qwen3ASR — Transformer-based encoder-decoder optimized for multilingual transcription - **Languages**: 55 Indic languages supported - **Format**: Transformers compatible (safetensors) - **Sampling Rate**: 16 kHz - **Precision**: FP16/FP32 supported - **Optimization**: Real-time inference capable with GPU acceleration ## Key Features - **Real-Time Performance**: Average RTFx of 300 enables real-time applications - **Verbatim Transcription**: Optimized for accurate, word-for-word transcription - **Multi-Domain Excellence**: Superior performance across conversational, broadcast, and read speech - **Multilingual Support**: 55 Indic languages with cross-lingual transfer learning - **HEEP-Curated Training**: Strategic entropy-based data selection for maximum information density ## Quick Start ### Install ```bash pip install qwen-asr[vllm] ``` ### Inference with vLLM (Recommended) ```python from qwen_asr import Qwen3ASRModel # Load model with vLLM backend asr = Qwen3ASRModel.LLM( model="bc7ec356/heep-indic", gpu_memory_utilization=0.8, max_new_tokens=4096, ) # Transcribe from file path results = asr.transcribe( audio="path/to/audio.wav", language="Hindi", ) print(results[0].text) print(results[0].language) ``` ### Inference with Transformers ```python import torch from qwen_asr import Qwen3ASRModel # Load model with Transformers backend asr = Qwen3ASRModel.from_pretrained( "bc7ec356/heep-indic", dtype=torch.bfloat16, device_map="cuda:0", ) # Transcribe results = asr.transcribe( audio="path/to/audio.wav", language="Hindi", ) print(results[0].text) ``` ### Batch Transcription ```python # Transcribe multiple files at once results = asr.transcribe( audio=["audio1.wav", "audio2.wav", "audio3.wav"], language=["Hindi", "Tamil", "Bengali"], ) for r in results: print(f"[{r.language}] {r.text}") ``` ### Auto Language Detection ```python # Pass language=None to auto-detect results = asr.transcribe( audio="path/to/audio.wav", language=None, ) print(f"Detected: {results[0].language}") print(f"Text: {results[0].text}") ``` ### Streaming Transcription (vLLM only) ```python import numpy as np import soundfile as sf from qwen_asr import Qwen3ASRModel asr = Qwen3ASRModel.LLM( model="bc7ec356/heep-indic", gpu_memory_utilization=0.8, max_new_tokens=4096, ) # Load audio wav, sr = sf.read("path/to/audio.wav", dtype="float32") # Initialize streaming state state = asr.init_streaming_state( language="Hindi", chunk_size_sec=2.0, unfixed_chunk_num=2, unfixed_token_num=5, ) # Feed audio in 1-second chunks step = sr # 1 second of samples for pos in range(0, len(wav), step): chunk = wav[pos : pos + step] asr.streaming_transcribe(chunk, state) print(f"Partial: {state.text}") # Finalize asr.finish_streaming_transcribe(state) print(f"Final: {state.text}") ``` ### NumPy Array Input ```python import numpy as np # From a numpy array + sample rate audio_array = np.random.randn(16000).astype(np.float32) # 1 second at 16kHz results = asr.transcribe( audio=(audio_array, 16000), language="English", ) ``` ## Performance Optimization Tips - **GPU Acceleration**: Use `device="cuda"` for significantly faster inference - **Precision**: Set `torch_dtype=torch.float16` for optimal speed on modern GPUs - **Language Specification**: Specify language code when known to improve accuracy and speed ## Acknowledgments HEEP Universal was developed using the HEEP framework for entropy-based data curation. We thank the open-source community for providing foundational tools that make this work possible. ## Citation If you use this model in your research, please cite: ```bibtex @article{anonymous2026heep, title={HEEP: High Entropy Exponential Pruning for State-of-the-Art ASR Through Strategic Data Curation}, author={Anonymous}, journal={Under Review}, year={2026} } ```