Qwen2-Audio-7B-Transcription

Qwen2-VL-Audio-Adapter

Multimodal Fusion: Integrating Whisper Audio Encoder with Qwen2-VL for Speech Recognition

Achieves 11.3% WER on the full official LargeScaleASR test partition (n=8,086); see full evaluation below.

🎯 Performance Highlights

Evaluation Context: WER and CER are the only directly computed metrics, measured on a 50-sample validation split — the head of the training stream (data/stage2_full/eval.json), not a held-out test set (60% exact-match rate). These 50 samples are not guaranteed disjoint from the training set, so treat the figure as a validation-split estimate. A separate blind manual audit of 100 samples from the SpeechBrain test partition characterized label noise (see below).

🧪 Test-Partition Result (Full Official Test Set)

Measured on the full official LargeScaleASR test partition (dataset revision 0e84cdb9e4b826afaabca5d33ec9453b11aacef3), n = 8,086 of 8,087 (1 excluded: test_6343, empty output). This is a held-out test set, distinct from the 50-sample validation split above.

95% CI (corpus): WER [10.84, 11.85], CER [6.32, 7.12].

A manual audit of 100 samples from this partition, all 100 reviewed against the audio, found 36 where the model's transcription was correct and the reference was wrong. The audited samples fall within this test partition. The reported figure is therefore an upper bound on true error rate.

Novel Finding: In the 100-sample manual audit, the model corrected ground-truth annotations in 36 of 100 samples — the majority of all disagreements between model and reference — demonstrating context-aware semantic reasoning. Note this is a sample-level rate from manual review, not a measured WER.

🏗️ Architecture

┌─────────────────────────────────────────────────┐
│  Whisper-Large-v3-Turbo Encoder (Frozen)        │
│  ~640M params → 1280-dim audio features         │
└────────────────┬────────────────────────────────┘
                 │
                 ↓
┌─────────────────────────────────────────────────┐
│  Audio Projector (Trainable)                    │
│  Linear: 1280 → 3584 dims (4.6M params)         │
└────────────────┬────────────────────────────────┘
                 │
                 ↓
┌─────────────────────────────────────────────────┐
│  Qwen2-VL-7B LLM (QLoRA Fine-tuned)             │
│  7B params with rank-64 LoRA adapters           │
└─────────────────────────────────────────────────┘

🔬 Rigorous Audit: Label Noise & Semantic Bias

To validate model quality on truly unseen data, I conducted a blind manual audit of 100 samples from the SpeechBrain test partition.

🔎 Audit Visualizer

1. Label Noise & Entity Resolution The model (Green) correctly identified "Mr. Šefčovič" (Maroš Šefčovič, EU Commissioner), correcting the ground truth "Mr. Efovi" (Red).

2. Semantic Bias & Long-Range Context The model "hallucinated" the word "Malta" (Green) in the first sentence because it attended to the context provided later in the audio, proving editorial reasoning.

Quantitative Analysis (N=100)

🧪 Training Infrastructure

• GPUs: Stage 1: 1× NVIDIA A100; Stage 2: 1× NVIDIA A6000 — single GPU per stage (no distributed training)
• Training time: ~18 GPU-hours total across both stages
• Framework: HuggingFace Transformers (custom fork) + PEFT + BitsAndBytes; BFloat16 + FlashAttention-2; Stage 2 uses 4-bit NF4 quantization (QLoRA)
• A rented H100 was used only for audit inference, not training.

💻 Usage

Important: This model requires a modified transformers library (included in the repo files).

Installation

Method 1: Git Clone (Recommended)

# Clone the model repo (includes transformers fork)
git clone https://huggingface.co/kulsoom-abdullah/Qwen2-VL-Audio-Adapter
cd Qwen2-VL-Audio-Adapter

# Install dependencies
pip install torch transformers librosa soundfile accelerate

Basic Inference

import sys
import torch
import librosa

# Load modified transformers from repo
sys.path.insert(0, "./transformers_fork/src")

from transformers import (
    Qwen2VLForConditionalGeneration,
    AutoTokenizer,
    WhisperFeatureExtractor
)

# Load model
model = Qwen2VLForConditionalGeneration.from_pretrained(
    "kulsoom-abdullah/Qwen2-VL-Audio-Adapter",
    torch_dtype=torch.bfloat16,
    device_map="auto",
    trust_remote_code=True
)

tokenizer = AutoTokenizer.from_pretrained(
    "kulsoom-abdullah/Qwen2-VL-Audio-Adapter",
    trust_remote_code=True
)

feature_extractor = WhisperFeatureExtractor.from_pretrained(
    "openai/whisper-large-v3-turbo"
)

# Load and prepare audio
audio_path = "your_audio.wav"
y, sr = librosa.load(audio_path, sr=16000, mono=True)
inputs = feature_extractor(y, sampling_rate=16000, return_tensors="pt")
input_features = inputs.input_features.to(model.device).to(torch.bfloat16)

# Build prompt
AUDIO_TOKEN_ID = 151657
NUM_AUDIO_TOKENS = 1500
audio_tokens = [AUDIO_TOKEN_ID] * NUM_AUDIO_TOKENS
input_ids_audio = torch.tensor([audio_tokens], device=model.device)

p1 = tokenizer.encode("<|im_start|>user\n<|audio_bos|>", add_special_tokens=False, return_tensors="pt").to(model.device)
p2 = tokenizer.encode("<|audio_eos|>\nTranscribe this audio.<|im_end|>\n<|im_start|>assistant\n", add_special_tokens=False, return_tensors="pt").to(model.device)
input_ids = torch.cat([p1, input_ids_audio, p2], dim=1)

# Generate
with torch.no_grad():
    generated_ids = model.generate(
        input_ids=input_ids,
        input_features=input_features,
        max_new_tokens=128
    )

print(tokenizer.decode(generated_ids[0][input_ids.shape[1]:], skip_special_tokens=True))

📝 Citation

@misc{qwen2-vl-audio-adapter,
  author = {Kulsoom Abdullah},
  title = {Qwen2-VL-Audio-Adapter: Multimodal Projection Alignment for Speech Recognition},
  year = {2026},
  publisher = {HuggingFace},
  howpublished = {\url{https://huggingface.co/kulsoom-abdullah/Qwen2-VL-Audio-Adapter}}
}

📄 License

Apache 2.0 (inherits from Qwen2-VL and Whisper)

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Kulsoom Abdullah | GitHub