---
license: cc-by-nc-4.0
tags:
  - speech-restoration
  - speech-enhancement
  - telephony
  - call-center
  - w2v-bert
  - lora
  - sidon
  - dac
base_model: facebook/w2v-bert-2.0
pipeline_tag: audio-to-audio
---

# CallEnhancer — Call-Centre / Telephony Speech Restoration

Restore Multi-lingual **narrowband, codec'd, noisy call-centre / telephony speech** (e.g. 8 kHz G.711/GSM phone
audio) to **clean 48 kHz**.

## Benchmark — CER

Character Error Rate (Whisper large-v3, **lower is better**) on a held-out **private call-centre
test set**. CallEnhancer restores telephony-degraded speech so an ASR reads it far more accurately
than the raw input or general-purpose speech enhancers.

![CER on a private call-centre test set](assets/callcentre_cer.png)

| model | CER % ↓ |
|---|---:|
| degraded (original) | 71.48 |
| voicefixer | 58.27 |
| resemble-enhance | 54.96 |
| sidon-v0.1 | 41.06 |
| **CallEnhancer-small** (this repo, open source) | **38.91** |
| **CallEnhancer-base** | **19.01** |

This repository hosts **CallEnhancer-small** (open source); **CallEnhancer-base** is the larger model.

## Quick start — infer from the HF checkpoint

```bash
pip install torch torchaudio "transformers>=4.56" "descript-audio-codec>=1.0.0" soundfile "huggingface_hub[cli]"

# pull the CLI + the two slim checkpoints from the Hub
hf auth login   # private repo: log in first (or export HF_TOKEN=hf_...)
hf download Scicom-intl/CallEnhancer \
    infer_callcentre.py fe_callcentre/fe_adapter_full.pt \
    decoder_callcentre/decoder_only.pt --local-dir CallEnhancer

cd CallEnhancer && python infer_callcentre.py \
    --input your_call.wav --out-dir out \
    --fe-adapter fe_callcentre/fe_adapter_full.pt \
    --decoder decoder_callcentre/decoder_only.pt \
    --chunk 0 --device cuda   # --chunk 0 = NO chunking (default single pass); --device cpu if no GPU
# -> out/your_call_restored48k.wav (clean 48 kHz)  +  out/your_call_orig48k.wav (A/B)
```

Prefer Python (load weights from the Hub with `hf_hub_download`)? See **[Python](#python-pull-weights-from-the-hub)** below.

> **Status:** CallEnhancer-small is fully trained and restores real 8 kHz call-centre audio well.

## Files

Use the **current-run** checkpoints under `fe_callcentre/` and `decoder_callcentre/`:

| path | role | size |
|---|---|---|
| `fe_callcentre/fe_adapter_full.pt` | **FE adapter (inference)** — 144 tensors: 96 LoRA + 48 trained `output_dense` biases | ~63 MB |
| `decoder_callcentre/decoder_only.pt` | **decoder (inference)** — 188M DAC decoder | ~0.75 GB |
| `fe_callcentre/last.pt`, `decoder_callcentre/last.pt` | raw checkpoints (resume training) | ~2.5 / 2.8 GB |
| `infer_callcentre.py` | inference CLI (below) | — |

For inference you only need the two slim files + `infer_callcentre.py`. *(Root-level
`fe_adapter_full.pt` / `decoder_only.pt` are from an earlier run and are superseded.)*

## End-to-end example (straight from HuggingFace)

```bash
pip install torch torchaudio "transformers>=4.56" "descript-audio-codec>=1.0.0" soundfile "huggingface_hub[cli]"

# pull the CLI + the two slim checkpoints, straight from this repo
hf auth login   # private repo: log in first (or export HF_TOKEN=hf_...)
hf download Scicom-intl/CallEnhancer \
    infer_callcentre.py \
    fe_callcentre/fe_adapter_full.pt \
    decoder_callcentre/decoder_only.pt \
    --local-dir CallEnhancer
cd CallEnhancer

# restore your audio end-to-end
python infer_callcentre.py \
    --input your_call.wav \
    --out-dir out \
    --fe-adapter fe_callcentre/fe_adapter_full.pt \
    --decoder   decoder_callcentre/decoder_only.pt \
    --chunk 0 --device cuda   # --chunk 0 = NO chunking (single straight pass, default); --device cpu if no GPU
```

Outputs:
- `out/your_call_restored48k.wav` — the restored **clean 48 kHz** speech.
- `out/your_call_orig48k.wav` — the input, naively upsampled to 48 kHz (no model), for an A/B listen.

`--input` accepts a **file or a directory** (`.wav/.flac/.mp3/.ogg/.opus/.m4a`). Stereo (e.g.
agent/customer on separate channels) is restored per channel and recombined.

### Python (pull weights from the Hub)

```python
import numpy as np, soundfile as sf, torch, torchaudio
from huggingface_hub import hf_hub_download
from transformers import AutoFeatureExtractor, Wav2Vec2BertModel
import dac

REPO, SSL, FE_SR, SR_OUT = "Scicom-intl/CallEnhancer", "facebook/w2v-bert-2.0", 16000, 48000
dev = torch.device("cuda" if torch.cuda.is_available() else "cpu")

ck = torch.load(hf_hub_download(REPO, "fe_callcentre/fe_adapter_full.pt"), map_location="cpu")
ad, scale = ck["adapter"], ck["lora_alpha"] / ck["r"]
fe = Wav2Vec2BertModel.from_pretrained(SSL, num_hidden_layers=ck.get("layers", 24), layerdrop=0.0)
sd = fe.state_dict()                                     # merge LoRA -> base (no peft needed)
for p in sorted({k[:-len(".lora_A.default.weight")] for k in ad if k.endswith(".lora_A.default.weight")}):
    sd[p+".weight"] = sd[p+".weight"].float() + scale * (ad[p+".lora_B.default.weight"].float() @ ad[p+".lora_A.default.weight"].float())
    if p+".base_layer.bias" in ad: sd[p+".bias"] = ad[p+".base_layer.bias"].to(sd[p+".bias"].dtype)
fe.load_state_dict(sd); fe.to(dev).eval()

dck = torch.load(hf_hub_download(REPO, "decoder_callcentre/decoder_only.pt"), map_location="cpu")
dec = dac.model.dac.Decoder(input_channel=1024, channels=dck.get("dec_channels", 3072), rates=[8,5,4,3,2])
dec.load_state_dict(dck["decoder"]); dec.to(dev).eval()
proc = AutoFeatureExtractor.from_pretrained(SSL)

@torch.no_grad()
def restore(path, out="restored48k.wav"):                # single straight pass
    x, sr = sf.read(path, dtype="float32"); x = x.mean(1) if x.ndim > 1 else x
    if sr != FE_SR: x = torchaudio.functional.resample(torch.from_numpy(x)[None], sr, FE_SR)[0].numpy()
    x = x / (np.abs(x).max() + 1e-9) * 0.95
    feats = {k: v.to(dev) for k, v in proc(x, sampling_rate=FE_SR, return_tensors="pt").items()}
    y = dec(fe(**feats).last_hidden_state.transpose(1, 2)).squeeze().float().cpu().numpy()
    sf.write(out, y / (np.abs(y).max() + 1e-9) * 0.97, SR_OUT); print("wrote", out)

restore("your_call.wav")   # <-- your own telephony/call-centre audio
```