Hueco #3 / 3c (CLI). NewRemuxSource now copies the video for any
browser-decodable codec: h264, or HEVC/AV1 when the web says the device
decodes them (caps). HEVC is muxed with -tag:v hvc1 (Apple requirement),
and non-aac audio (ac3/eac3/dts) is transcoded to aac while the video is
still copied (ActionRemuxAudio) — this covers the very common h264+ac3 mkv.
Startup instrumentation for time-to-first-frame diagnosis:
- remux branch logs [probe=.. spawn=..]
- transcodeSource logs 'first fMP4 bytes after ..' (ffmpeg → first output)
- serveGrowing logs reads that block >250ms (client seeking ahead of the
live edge) + the first read's offset vs produced/estimated size.
Verified: caps gate (hls without caps, remux with), hvc1 retag (ffprobe of
the /stream output = hevc/hvc1), HEVC playback confirmed on a real iPhone
Safari over Tailscale. LAN timeline: probe 16ms, spawn 1ms, first byte
201ms, no serveGrowing blocks.
Agent side of 3b: serve a growing ffmpeg `-c copy` remux (mkv h264/aac →
fragmented MP4) over /stream with no video re-encode. Dormant until the web
sends PlayMethod="remux" (3b-ii), so this commit changes no live behavior.
- GrowingSource interface + transcodeSource already satisfies it; estimate is
the source file size for copy actions (≈ remux output) vs bitrate×duration
for real transcodes.
- NewRemuxSource: ffmpeg -c copy → growing fMP4 temp, returned as GrowingSource.
- StreamServer.SetGrowingFile + serveGrowing: manual Range responder for a
growing source (http.ServeContent needs a fixed size). 206 with an estimated
total in Content-Range; chunked body while not final (never promise bytes a
running remux might not produce); exact Content-Length once final. Blocks via
ReadAt for not-yet-produced bytes; forward seek waits, backward seek instant.
- daemon OnStreamSession: PlayMethod=="remux" → NewRemuxSource + SetGrowingFile
+ MarkSessionReady (after the ffmpeg check; copy still needs ffmpeg).
- Tests: parseByteRange + serveGrowing (full/offset/bounded/estimate/HEAD/416).
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.
1. -profile:v main + -level:v 4.0 to avoid Chrome's HW decoder path that
failed with "VaapiWrapper: failed initializing for h264 high" on Linux.
2. setparams to rewrite HDR HEVC color metadata to SDR Rec.709 so browsers
don't reject wide-gamut output.
3. serveRange caps `want` by estimated final size (not current). ReadAt
blocks until ffmpeg catches up — that's the right behaviour. Returning
RangeEnd inmediato was making the browser abort with "Format error".
4. Debug log on every range_req.
Source files in HEVC, AV1, AC3, DTS, EAC3, etc. now transcode through ffmpeg
to fragmented MP4 (h264 + aac) on-the-fly when the browser would otherwise
play silent black. Decision matrix lives in engine.DecideAction:
passthrough → remux → audio-transcode → full video-transcode.
Architecture — temp file + growing-size source:
- engine.streamSource interface abstracts byte source. Two impls:
* diskFileSource: passthrough when codecs are already browser-friendly.
* transcodeSource: spawns ffmpeg writing to a /tmp/tc-stream-*.mp4 file.
A ticker polls file size and wakes blocked ReadAt callers as ffmpeg
produces output. Estimate of final size (bitrate × duration) is
announced over the wire so the browser's scrubber has something to
anchor on.
- dataChannelPump now reads from streamSource instead of *os.File. HELLO
carries Transcoding=true + an estimated total size; Seekable=true (we
read random-access from the temp file even while writing).
- Transcoder runtime resolved per session by buildTranscodeRuntime in
cmd/daemon: ffmpeg/ffprobe path lookup + HWAccel auto-detection
(NVENC/QSV/VAAPI/VideoToolbox).
- New [downloads.transcode] TOML section: enabled (default true), hw_accel
(auto), preset (veryfast), video_bitrate (5M), audio_bitrate (192k),
max_height (optional downscale), max_concurrent (safety cap).
Falls back to passthrough if ffprobe is missing, fails, or codecs are
already browser-friendly. tmp file is cleaned up on session shutdown.
