GGUF quantizations of the CED family (Consistent Ensemble Distillation, Xiaomi) - SOTA-tier audio-tagging models that classify everyday sounds (baby cry, footsteps, glass breaking, alarms, dog bark,...) into the 527-class AudioSet ontology. These files run with ced.cpp, a standalone C++/ggml port (no Python, no PyTorch at inference), and with LocalAI via the ced backend. Converted from the mispeech/ced- checkpoints (Apache-2.0). CED is a plain AST/DeiT Vision Transformer over a log-mel spectrogram; the port is numerically equal to the PyTorch reference. One self-contained GGUF per size + quant (config, 527 labels, and the mel filterbank/window are all embedded). Pick by your accuracy/size…
campplus is an open-weight model for audio classification from losCrossos, released under Apache License 2.0. Its published files total 28.5 MB.
This model is part of the FunASR ecosystem — one industrial-grade open-source / event · LLM-ASR. A Star really helps the project (and keeps you updated): Speaker Verification & Diarization — identify and distinguish speakers in audio.
Model Card
By losCrossos, published under apache-2.0, revision 185ffb07eb4d.
This model is part of the FunASR ecosystem — one industrial-grade open-source / event · LLM-ASR. A Star really helps the project (and keeps you updated): Speaker Verification & Diarization — identify and distinguish speakers in audio. CAM++ is a speaker embedding model used for speaker verification (is this the same person?) and speaker diarization (who spoke when?). - Works with FunASR pipeline via spkmodel parameter
Read losCrossos's full model card
CAM++
Speaker Verification & Diarization — identify and distinguish speakers in audio.
CAM++ is a speaker embedding model used for speaker verification (is this the same person?) and speaker diarization (who spoke when?).
Quick Start
from funasr import AutoModel
# Speaker diarization as part of ASR pipeline
model = AutoModel(
model="funasr/paraformer-zh",
hub="hf",
vad_model="funasr/fsmn-vad",
punc_model="funasr/ct-punc",
spk_model="funasr/campplus",
device="cuda",
)
result = model.generate(input="meeting.wav")
# Each sentence has a speaker label
for sentence in result[0]["sentence_info"]:
print(f"[Speaker {sentence['spk']}] {sentence['text']}")
Features
- Speaker embedding extraction
- Speaker verification (same/different speaker)
- Speaker diarization (multi-speaker segmentation)
- Works with FunASR pipeline via
spk_modelparameter
Model Details
| Property | Value |
|---|---|
| Architecture | CAM++ (Class-Aware Multi-scale) |
| Embedding Dim | 192 |
| Sample Rate | 16kHz |
Links
- GitHub: FunASR
- Docs: modelscope.github.io/FunASR
Identity and Version
- Repository
- loscrossos/campplus
- Publisher
- losCrossos
- Task
- Audio classification
- Modality
- Audio
- Library
- funasr
- Parameters
- Not stated by the source
- Languages
- zh, en
- Revision
- 185ffb07eb4d078e778815e2534d5edec1175e76
- First published
- 2026-10-13
- Last updated
- 2026-10-13
Files and Weights
8 files, 28.5 MB in total. The weights are 1 file totalling 28.0 MB in bin.
Every file
| File | Type | Size | SHA-256 |
|---|---|---|---|
| campplus_cn_common.bin | Weights | 28.0 MB | 3388cf5fd349 |
| config.yaml | Configuration | 537 B | — |
| configuration.json | Configuration | 581 B | — |
| README.md | Documentation | 2.0 KB | — |
| quickstart.md | Documentation | 1.5 KB | — |
| dingding.jpg | Other | 184.0 KB | — |
| structure.png | Other | 286.0 KB | — |
| .gitattributes | Repository | 1.5 KB | — |
License and Download
- License
- apache-2.0
- Access
- Open weights, no gate
- Download size
- 28.0 MB
Released by losCrossos through its official repository on Hugging Face. Read the license.
Memory Requirements
| Precision | Weights in memory |
|---|---|
| As published | 28.0 MB |
Weights only, from the published parameter count; the key-value cache and runtime add to this.
Questions About campplus
Can I use campplus commercially?
Yes. campplus is released under Apache License 2.0. The Apache License 2.0 is a permissive open-source license. It permits commercial use, modification and redistribution. It requires keeping the license and copyright notices and any NOTICE file, stating significant changes, and it includes an express patent grant from contributors.
Similar Models
This is a spoken language recognition model trained on the VoxLingua107 dataset using SpeechBrain. The model uses the ECAPA-TDNN architecture that has previously been used for speaker recognition. However, it uses more fully connected hidden layers after the embedding layer, and cross-entropy loss was used for training. We observed that this improved the performance of extracted utterance embeddings for downstream tasks. The system is trained with recordings sampled at 16kHz (single channel). The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling classifyfile if needed. The model can classify a speech utterance according to the language…
Abstract: The recognition of accented speech still remains a dominant problem in Automatic Speech Recognition (ASR) systems. We approach the classification of accented English speech through the Emphasized Channel Attention, Propagation and Aggregation Time Delay Neural Network (ECAPA-TDNN) architecture which has been shown to perform well on a variety of speech tasks. Three models are proposed: one trained from scratch, another two models (one using data augmentation and a baseline model) fine-tuned from the checkpoints of speechbrain/spkrec-ecapa-voxceleb (VoxCeleb). Our results show that the model fine-tuned with data augmentation yield the best results. Most of the misclassifications…
The development log of our Music Audio Pre-training (m-a-p) model family: - 02/06/2023: arxiv pre-print and training codes released. - 17/03/2023: we release two advanced music understanding models, MERT-v1-95M and MERT-v1-330M, trained with new paradigm and dataset. They outperform the previous models and can better generalize to more tasks. - 14/03/2023: we retrained the MERT-v0 model with open-source-only music dataset MERT-v0-public - 29/12/2022: a music understanding model MERT-v0 trained with MLM paradigm, which performs better at downstream tasks. - 29/10/2022: a pre-trained MIR model music2vec trained with BYOL paradigm. Here is a table for quick model pick-up: The m-a-p models…
The development log of our Music Audio Pre-training (m-a-p) model family: - 02/06/2023: arxiv pre-print and training codes released. - 17/03/2023: we release two advanced music understanding models, MERT-v1-95M and MERT-v1-330M, trained with new paradigm and dataset. They outperform the previous models and can better generalize to more tasks. - 14/03/2023: we retrained the MERT-v0 model with open-source-only music dataset MERT-v0-public - 29/12/2022: a music understanding model MERT-v0 trained with MLM paradigm, which performs better at downstream tasks. - 29/10/2022: a pre-trained MIR model music2vec trained with BYOL paradigm. Here is a table for quick model pick-up: The m-a-p models…
This model combines the SpeechBrain ECAPA-TDNN speaker embedding model with an SVM classifier to predict speaker gender from audio input. The model was trained and evaluated on the VoxCeleb2, Mozilla Common Voice v10.0, and TIMIT datasets - Mozilla Common Voice v10.0 English validated test set: 92.3% accuracy The model was trained on VoxCeleb2 dataset: - Converted to WAV format, single channel, 16kHz sampling rate, 256 kp/s bitrate - Applied SileroVAD for voice activity detection, taking the first voiced segment You can install the package directly from GitHub: - Model was trained on celebrity voices from YouTube interviews - Performance may vary on different audio qualities or recording…
