Close the recurring "video has subtitles but the web player shows none" gap
with a source-agnostic pipeline:
- Discover EXTERNAL sidecar subs in the scan (Video.es.ass siblings + a Subs/
bundle), parse lang/forced/SDH from the filename, skip VobSub (.sub+.idx).
ffprobe-only scanning ignored these (ToonsHub/anime "MSubs" releases).
- Transcode sidecar charset -> UTF-8 before WebVTT (BOM/UTF-16/code-page by
language). Chinese SCRIPT matters: chs/sc -> GBK, cht/tc/big5 -> Big5
(decoding one as the other is mojibake).
- /sub now serves a standalone sidecar file (i=-1, p=file, &l=lang hint) and a
remote debrid URL (ffmpeg reads http, no local stat) — not just embedded
streams of a local file.
- probe.json emits a tokened vttUrl per TEXT track so torrent/debrid HLS streams
(never library-scanned) get subtitles too. Embedded index is counted among
embedded streams only, so -map 0:s:N stays aligned when sidecars are appended.
Tested against a real 347-file gallery: 26/26 sidecars and embedded ass/srt/
mov_text all extract to valid WebVTT; bitmap (pgs/dvd_subtitle) correctly stays
burn-in. Manual harness gated behind GALLERY_DIR.
Bitmap subs can't be served as WebVTT, so the user picks one and the daemon
re-encodes with it overlaid. HLSSessionConfig.BurnSubtitleIndex (*int, nil=no
burn) flows into the cache key + a -filter_complex graph:
[0✌️0]<vchain>[base];[0:s:N][base]scale2ref[sub][base2];[base2][sub]overlay[vout]
Overlay after the tonemap (SDR subs keep brightness); scale2ref fits the PGS
canvas to the output. Invalid/text/out-of-range index -> clean-encode fallback.
IsTextSubtitle now includes "text" (parity with the web classifier).
Reduces first-segment latency on cache MISS so the player doesn't sit on
"preparando sesión". Three independent levers:
1. ProbeFile memoised by (path, mtime, size) for 30 min — second play of
the same source skips ffprobe (1-3 s on 50+ GB MKVs).
2. HLS encoder presets biased for latency over quality:
- libx264 default veryfast → superfast (~15-20% faster, marginal
quality loss at 5-25 Mbps target bitrates).
- NVENC: -preset p4 -tune hq → -preset p3 -tune ll. First-segment
~0.8 s on RTX-class GPUs (was ~1.5 s).
- QSV: -preset medium → -preset veryfast (keeps look_ahead=0).
- VideoToolbox: adds -realtime 1 (was unset). Bitrate args still
drive rate control; -q:v dropped to avoid the silent conflict
where ffmpeg ignored it under -b:v.
3. Per-session log surfaces encoder + accel + preset so "first-start
was slow" complaints can be triaged from the journal alone.
Diagnostic helpers (DetectHWAccelDiagnostic + HWAccelDiagnostic) added
for future wiring into daemon startup / agent register; users today can
already inspect via `unarr probe-hwaccel`.
Web: AgentsTab profile page now shows the agent's chosen encoder
(amber if software libx264, green if HW) plus the transcode-resolution
cap. Hidden for pre-0.9.9 agents that haven't reported hwAccel.
Drops the custom WebRTC DataChannel pipeline + pion deps + WSS signaling
client + wire framing. Every in-browser playback now uses HLS over HTTP
from the daemon (Tailscale/LAN/UPnP). Browser P2P never re-enabled.
Wire renames (incompatible with web < 2026-05-26): agent.WebRTCSession
=> agent.StreamSession, SyncResponse.WebRTCSessions (JSON: webrtcSessions)
=> StreamSessions (JSON: streamSessions). MIN_AGENT_VERSION is bumped
to 0.9.4 on the web side so older agents see an upgrade card.
Also fixes the libx264 'VBV bitrate > level limit' abort by clamping
the encoder bitrate to the effective output height instead of the
requested label (carried over from the prior 0.9.3 unreleased work).
The seed_file vertical (mode=seed_file handler + engine.SeedFile) was
retired with the in-browser P2P player. [downloads.webrtc] config block
deleted; existing TOML files with the section still parse fine.
Introduces an HLS-over-HTTP path as Plan B for in-browser streaming. The
WebRTC + MSE pipeline keeps working untouched; the new path is selected
when the backend sets transport="hls" on a streaming session.
Daemon scope:
- engine/hls.go: HLSSession + HLSSessionRegistry. Spawns ffmpeg with
-f hls -hls_segment_type fmp4 + force_key_frames aligned with 4 s
segments. Pre-renders master + media playlists from the probe duration
so the browser knows the total timeline before any segment exists,
fixing seek/duration/pause/multi-track issues seen with the live fMP4
pipe.
- engine/probe.go: enumerate every audio + subtitle track instead of
collapsing to a single default audio track.
- engine/stream_server.go: route /hls/<id>/{master.m3u8,video/...,
subs/...} to the matching session. Emit a synthesised single-VTT
subtitle playlist per text track; bitmap subs (PGS/DVB) skip silently.
- cmd/daemon.go: branch on WebRTCSession.Transport == "hls" to register
an HLS session instead of running the legacy DataChannel pump.
- agent/types.go: WebRTCSession.Transport + AudioIndex fields.
Backend + web sides land in a follow-up commit.
