luke eaedc4214b Reduce live caller latency and improve reliability ...

- Replace per-callback async task spawning with persistent queue-based sender
- Buffer host mic to 60ms chunks (was 21ms) to reduce WebSocket frame rate
- Reduce server ring buffer prebuffer from 150ms to 80ms
- Reduce browser playback jitter buffer from 150ms to 100ms

Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>

2026-02-05 16:47:17 -07:00

backend

Reduce live caller latency and improve reliability

2026-02-05 16:47:17 -07:00

frontend

Reduce live caller latency and improve reliability

2026-02-05 16:47:17 -07:00

tests

Update tests for CallerService and browser caller format

2026-02-05 15:53:41 -07:00

.gitignore

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

audio_settings.json

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

CLAUDE.md

Add CLAUDE.md with project config and Gitea setup

2026-02-04 23:21:50 -07:00

download_sounds.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

generate_callers.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

generate_sounds.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

publish_episode.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

radio_show.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

radio_simple.py

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

requirements-web.txt

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

run.sh

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

test.html

Initial commit: AI Radio Show web application

2026-02-04 23:11:20 -07:00

S

Description

AI Radio Show - web-based podcast production with multiple TTS providers

MIT 18 MiB

Languages

Python 80.1%

JavaScript 6.8%

HTML 6.4%

CSS 3.9%

Lua 2.4%

Other 0.4%