Juergen Kunz
f3e1c96872
Full-featured SIP router with multi-provider trunking, browser softphone via WebRTC, real-time Opus/G.722/PCM transcoding in Rust, RNNoise ML noise suppression, Kokoro neural TTS announcements, and a Lit-based web dashboard with live call monitoring and REST API.
13 lines
1.0 KiB
Markdown
13 lines
1.0 KiB
Markdown
# Ideas / Future Improvements
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## nnnoiseless (RNNoise) Denoiser Improvements
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### VAD-gated passthrough
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`process_frame()` returns an `f32` VAD probability (0.0-1.0). Currently ignored. Use it to skip denoising when VAD is low — prevents the model from suppressing non-speech audio (hold music, DTMF tones, IVR prompts).
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### Pre-warm denoiser on session creation
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The first `process_frame()` call on a fresh `DenoiseState` produces fade-in artifacts (documented behavior). Feed a silent 480-sample frame during `TranscodeState::new()` so the first real audio frame gets a warmed-up RNN state.
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### Custom telephony-trained RNNoise model
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nnnoiseless supports loading custom `.rnn` model files via `RnnModel::from_bytes()` / `RnnModel::from_static_bytes()`. The default model is trained on general audio. A model trained specifically on telephony noise profiles (codec artifacts, line noise, echo residual) would perform better. Models from https://github.com/GregorR/rnnoise-models can be converted with `train/convert_rnnoise.py`.
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