- Add host mic noise reduction (afftdn + anlmdn)
- Add phone EQ bandpass on caller stem
- Mute music during ads with 2s lookahead/tail
- Increase ducking release to 3s to reduce pumping
- Add Inworld voice mappings for all regular callers
- Recording toggle endpoint, stem sync fixes
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Returning callers now keep their voice across sessions (stored in regulars.json)
- Backfilled voice assignments for all 11 existing regulars
- Discord button on homepage + link in all page footers
- REC and On-Air buttons now toggle together (both directions)
- Fixed host mic double-stream bug (stem_mic vs host_stream conflict)
- SEO: JSON-LD structured data on episode + how-it-works pages
- SEO: noscript fallbacks, RSS links, twitter meta tags
- Episode 9 transcript and sitemap update
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Re-label all 8 episode transcripts with LUKE:/CALLER: speaker labels
using LLM-based diarization (relabel_transcripts.py)
- Add episode.html transcript page with styled speaker labels
- Update publish_episode.py to generate speaker-labeled transcripts
and copy to website/transcripts/ for Cloudflare Pages
- Add SVG favicon with PNG fallbacks
- Fix CPU issue: tie host audio stream to on-air toggle, not per-caller
- Update how-it-works page with post-production pipeline info
- Add transcript links to episode cards in app.js
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- On-air toggle uploads status.json to BunnyCDN + purges cache, website
polls it every 15s to show live ON AIR / OFF AIR badge
- Publish script downloads Castopod's copy of audio for CDN upload
(byte-exact match), removes broken slug fallback, syncs all episode
media to CDN after publishing
- Fix f-string syntax error in publish_episode.py (Python <3.12)
- Enable CORS on BunnyCDN pull zone for json files
- CDN URLs for website OG images, stem recorder bug fixes, LLM token
budget tweaks, session context in CLAUDE.md
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
New stem recording system captures 5 time-aligned WAV files (host, caller,
music, sfx, ads) during live shows. Standalone postprod.py processes stems
into broadcast-ready MP3 with gap removal, voice compression, music ducking,
and EBU R128 loudness normalization.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Music crossfade: smooth 3-second blend between tracks instead of hard stop/start
- Emotional detection: analyze host mood from recent messages so callers adapt tone
- AI caller summaries: generate call summaries with timestamps for show history
- Returning callers: persist regular callers across sessions with call history
- Session export: generate transcripts with speaker labels and chapter markers
- Caller screening: AI pre-screens phone callers to get name and topic while queued
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add dedicated ad playback system (no loop, own channel)
- Ad channel defaults to 11, saved/loaded with audio settings
- Separate play_ad/stop_ad methods and API endpoints
- Frontend stop button now calls /api/ads/stop instead of stopMusic
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Log chunk count and peak audio level on recording stop
- Add null check on _recorded_audio in callback
- Small delay after stopping piggybacked recording for callback to finish
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Mute host mic forwarding while TTS is streaming to prevent interleaving
both audio sources into the same playback buffer
- Replace nearest-neighbor downsampling with box-filter averaging on both
server (host mic) and browser (caller mic) for anti-aliased resampling
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- 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>
When a live caller is on air, the host stream already has an InputStream
open. Opening a second one for push-to-talk recording causes a conflict.
Now recording piggybacks on the host stream callback instead.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Reduce capture chunk from 4096 to 640 samples (256ms → 40ms)
- Replace BufferSource scheduling with AudioWorklet playback ring buffer
- Add 80ms jitter buffer with linear interpolation upsampling
- Reduce host mic and live caller stream blocksizes from 4096/2048 to 1024
- Replace librosa.resample with numpy interpolation in send_audio_to_caller
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Use persistent callback-based output stream instead of opening/closing per chunk
- Replace librosa.resample with simple decimation in real-time audio callbacks
- Move host stream initialization to background thread to avoid blocking
- Change live caller channel default to 9
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
