ReDimNet2+: Speaker Verification Model
ReDimNet2+ is a robustness-oriented adaptation of the compact ReDimNet2-B6 backbone for Automatic Speaker Verification (ASV), trained for robustness to channel and codec mismatch (compression, telephony codecs, noise, reverberation). The model uses a ReDimNet2 backbone with a Mel-spectrogram frontend and produces 192-dimensional speaker embeddings.
- Code (training, eval, Docker submission pipeline): github.com/lab260ru/redimnet2-plus
- Paper: ReDimNet2+ for Robust Speaker Verification under Channel and Codec Shift (manuscript submitted to ICASSP 2027)
Results
Evaluated with random 4-second windows and cosine similarity on L2-normalized embeddings (VoxCeleb1, EER %; EERp pools O/E/H; EERph is a 26-condition codec/waveform stress test). See the paper for the full protocol and ablations.
| System | EERo | EERe | EERh | EERp | EERph |
|---|---|---|---|---|---|
| ReDimNet2 baseline (public ckpt.) | 1.601 | 1.725 | 3.039 | 2.420 | 7.214 |
| WeSpeaker CAM++ (best evaluated WeSpeaker ckpt.) | 0.787 | 0.928 | 1.824 | โ | โ |
| ReDimNet2+ LMFT (this checkpoint) | 0.351 | 0.523 | 1.055 | 0.824 | 1.991 |
Model Architecture
- Backbone: ReDimNet2-B6
- Embedding Dimension: 192
- Input: Mel-spectrograms (64 mel bins, 10ms hop)
- Sample Rate: 16 kHz
- Frontend: Log-Mel spectrogram with 64 mel bands
- Pooling: ASTP (Attentive Statistics Aware Pooling)
- Normalization: L2 normalization on embeddings
Usage
Installation
pip install torch torchaudio soundfile omegaconf
Loading the Model
import torch
from huggingface_hub import hf_hub_download
from omegaconf import OmegaConf
# Download model weights
model_path = hf_hub_download("lab260/redimnet2-plus", "pytorch_model_fsdp.bin")
# Load checkpoint
state_dict = torch.load(model_path, map_location="cpu")
# Strip FSDP wrapper prefixes
state_dict = {
k.replace("_orig_mod.", "").replace("module.", ""): v
for k, v in state_dict.items()
}
# Model configuration
model_config = {
"encoder": {
"_target_": "asv.models.redimnet2.ReDimNet2Encoder",
"model_name": "b6",
"pretrained": False,
"train_type": "lm",
"dataset": "vox2",
"strict_load": True,
"model_overrides": {
"out_channels": 224,
"return_2d_output": False
}
},
"bridge": {
"_target_": "asv.models.bridge.IdentityBridge"
},
"classifier": {
"_target_": "asv.models.classifier.IdentityClassifier"
}
}
Extracting Embeddings
import torchaudio.transforms as T
def load_audio(path, sr=16000):
"""Load audio file."""
audio, sr = torchaudio.load(path)
if audio.shape[0] > 1:
audio = audio.mean(dim=0) # Convert to mono
return audio, sr
def compute_mel_spec(audio, sr=16000):
"""Compute mel-spectrogram."""
mel_transform = T.MelSpectrogram(
sample_rate=sr,
n_fft=512,
hop_length=160,
n_mels=64,
normalized=False
)
mel = mel_transform(audio.unsqueeze(0))
mel = torch.log(mel + 1e-6)
return mel
def extract_embedding(model, audio_path, device="cpu"):
"""Extract speaker embedding from audio."""
# Load and preprocess
audio, sr = load_audio(audio_path)
mel = compute_mel_spec(audio, sr)
# Add channel dim
if mel.ndim == 3:
mel = mel.unsqueeze(1)
mel = mel.to(device)
model = model.to(device)
model.eval()
with torch.no_grad():
embedding = model(mel)
embedding = torch.nn.functional.normalize(embedding, dim=1)
return embedding.cpu().squeeze(0)
# Example usage
embedding = extract_embedding(model, "audio.wav")
print(f"Embedding shape: {embedding.shape}") # [192]
Speaker Verification
def cosine_similarity(e1, e2):
"""Calculate cosine similarity between embeddings."""
return torch.dot(e1, e2).item()
def verify_speaker(model, audio1_path, audio2_path, threshold=0.79):
"""Verify if two audio samples are from the same speaker."""
emb1 = extract_embedding(model, audio1_path)
emb2 = extract_embedding(model, audio2_path)
similarity = cosine_similarity(emb1, emb2)
is_same = similarity >= threshold
return {
"similarity": similarity,
"is_same_speaker": is_same,
"confidence": abs(similarity - threshold)
}
Model Details
- File Format: PyTorch state_dict (binary)
- File Size: ~100 MB
- Framework: PyTorch 2.11+
- Key Prefix:
_orig_mod.(should be stripped during loading)
Training
Full recipe and code: github.com/lab260ru/redimnet2-plus. Summary:
- Initialization: public ReDimNet2-B6 checkpoint, backbone architecture unchanged.
- Data: 7 corpora, ~63.9k speakers, ~4.6M utterances, ~8,675 h โ VoxBlink2, VoxCeleb2, 3D-Speaker, CN-Celeb/CN-Celeb2, TidyVoice, KeSpeech.
- Augmentation: filters, additive noise (colored, MUSAN, RawBoost-style), simulated RIRs, speed perturbation, and codec simulation (MP3, Opus, AAC, FLAC, Vorbis, G.723.1, IMA ADPCM, G.722, A-law, ฮผ-law, Speex, AMR-NB at 8/16 kHz), plus SpecAugment/CutMix.
- Loss Function: SphereFace2 (one-vs-all)
- Optimizer: AdamW, BFloat16, Accelerate + FSDP on 6 GPUs, gradient clipping at 20
- Recipe: staged multi-corpus adaptation followed by large-margin fine-tuning (LMFT) on 6-second windows (margin 0.3)
This checkpoint (pytorch_model_fsdp.bin) is the final ReDimNet2+ LMFT model from the paper.
Retrieval reranking
The GitHub repo also ships a training-free, graph-based reranking stage (mean-chain + k-NN graph reranking with hubness correction) for open-set retrieval on top of these embeddings โ see submission/reranking/. It raises VoxBlink2-subset Pr@k from 0.7413 to 0.7687 without retraining.
Limitations
- Model expects 16 kHz audio input
- Maximum audio duration: 60 seconds
- EER roughly doubles from VoxCeleb1-E to -H (hard impostors), and the 26-condition codec/waveform stress test (EERph) stays above the clean metrics even after LMFT โ see the paper's limitations section
- External comparisons in the paper are shared-protocol, not equal-data comparisons
Citation
@unpublished{borodin2027redimnet2plus,
title = {{ReDimNet2+} for Robust Speaker Verification under Channel and Codec Shift},
author = {Borodin, Kirill and Kudryavtsev, Vasiliy and Mkrchian, Grach},
year = {2026},
note = {Manuscript submitted to ICASSP 2027}
}
License
This model is released under the Apache 2.0 license.
Contact
- Email: kborodin.research@gmail.com
- Telegram: @korallll_ai
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