## Speech Enhancement Standard (SE2.1)

The Speech Enhancement Standard (`SE2.1`) model restores and enhances voice quality for telephony audio. It takes 16kHz input and outputs high-fidelity 8kHz audio, correcting codec degradation, packet loss, reverb, and background noise — purpose-built for contact center and telephony environments with limited CPU budgets.

## Hear the Difference

### Various Accents

**Before** · Original audio  
**After** · Enhanced with SE Standard

### Codec Degradation

**Before** · Original audio  
**After** · Enhanced with SE Standard

### Packet Loss

**Before** · Original audio  
**After** · Enhanced with SE Standard

### Overlapping Speech

**Before** · Original audio  
**After** · Enhanced with SE Standard

### Background Noise

**Before** · Original audio  
**After** · Enhanced with SE Standard

## Key Features

### Removes Noise and Speech

Eliminates background noises and background speech while preserving the primary speaker.

### Restores Voice Fidelity

Restores foreground voice to high-fidelity, enhancing clarity, articulation, energy, and vocal presence.

### Corrects Degradation

Fixes codec degradation, packet loss from poor connections, reverb, and room acoustics.

### Low CPU Footprint

Designed for production environments with limited CPU resources.

## Specifications

**Model ID:** `SE2.1`  
**Category:** Speech Enhancement  
**Type:** Human ↔ Human

### 120ms

Streaming latency

### 16kHz → 8kHz

Input / Output sample rate

## Use Cases

### Contact Centers

Enhance agent-customer call quality in telephony environments with limited bandwidth.

### Telephony Systems

Restore voice clarity for 8kHz telephony pipelines degraded by compression and network conditions.

### IVR Systems

Improve audio quality for automated phone systems operating on narrowband audio.

### Voice Recording

Clean up and restore archived or low-quality voice recordings.

**Known Limitations**

- Speaker identity may occasionally differ slightly from the original.
- Although pronunciations are unchanged, there may be a subtle perception of an Americanized accent.

## Code Example

Create an audio processor with the SE Standard model:

```
audio_params = sanas_remote_sdk.AudioParams()
audio_params.modelName = "SE2.1"
audio_params.sampleRate = 16000

processor, create_result = sdk.CreateAudioProcessor(audio_params)
```

Input sample rate: 16 kHz. The model outputs 8 kHz optimized for telephony, but Sanas Cloud upsamples it to 16 kHz to match the input sample rate.

For full setup and initialization, see the [Quick Start →](https://developer.sanas.ai/Docs/Getting-Started/Quick-Start)

## Next Steps

[**Quick Start**](https://developer.sanas.ai/Docs/Getting-Started/Quick-Start)  
[**API Reference**](https://developer.sanas.ai/API-Reference/Overview)  
[**Processing Multiple Streams**](https://developer.sanas.ai/Docs/Tutorials/Processing-Multiple-Streams)