deps: update esbuild to 0.27.4, rollup to 4.60.0, vite to 7.3.1
- bump esbuild and all platform-specific binaries from 0.25.10 to 0.27.4
- bump rollup and all platform-specific binaries from 4.52.3 to 4.60.0
- bump vite from 7.1.7 to 7.3.1 (esbuild peer dep updated to ^0.27.0)
- bump dompurify from 3.2.7 to 3.3.3
web: improve cover image retry logic
- increase maxRetries default from 3 to 5
- reduce initial retryDelay from 1000ms to 500ms
- implement exponential backoff: delay = retryDelay * 2^(retryCount - 1)
- add detailed logging for retries and final failure
rust: minor cleanup
- remove unused imports (watch, Url, AudioSegment, SyncMarker)
- add tracing debug logs for cache file requests
- handle missing files gracefully with warning + client retry hint
- add #[allow(async_fn_in_trait)] where needed
Invalidate the cached OGG header immediately when restarting the FLAC encoder to prevent clients from receiving stale data.
Also improve the player source cleanup by draining the chunk receiver and adding a timeout when waiting for the emit task to finish, preventing potential hangs when the downstream pipeline is saturated.
Ajout de l'événement Position pour envoyer la position courante ~1/s pendant la lecture.
Modification de la gestion de la durée dans le registry pour ne pas écraser une durée déjà connue par la source (priorité à la source pour les flux radio).
Mise à jour du handler position_update pour utiliser update_duration au lieu de update_position, et suppression de la gestion de position_sec qui est maintenant gérée par PlayerEvent::Position.
Le serveur gère maintenant la position via les événements Position émis par PlayerSource, et la durée est uniquement utilisée comme fallback pour les sources sans durée connue.
Ajout de PlayerSource avec contrôle complet AVTransport (Play/Pause/Stop/Seek/LoadUri) dans le pipeline audio
- Introduit PlayerSource comme nœud source audio avec gestion complète du cycle de vie UPnP
- Implémente les commandes LoadUri, LoadNextUri, Play, Pause, Stop, Seek
- Gestion des transitions gapless avec TrackBoundary pour un bitstream propre
- Révision du pipeline audio pour utiliser PlayerSource au lieu de la logique de contrôle existante
- Mise à jour des handlers UPnP pour utiliser PlayerHandle au lieu du canal de contrôle
- Suppression des anciennes commandes PipelineControl et logique de gestion de source
- Ajout de gestion d'événements PlayerEvent pour mise à jour de l'état UPnP
- Migration vers StreamingOggFlacSink pour la diffusion audio
Migrate from single-client direct OGG-FLAC sink to multi-client broadcast streaming sink.
- Replace DirectOggFlacHandle with OggFlacStreamHandle for multi-client support
- Update sink implementation to use StreamingOggFlacSink with proper backpressure
- Change connection model from 'connect()' to 'subscribe()'
- Adjust stream handling to support independent client streams
- Update stream endpoint to always respond with 200 chunked instead of range requests
- Add tracing for stream lifecycle events
- Reduce OGG channel capacity to strict backpressure (1)
- Add Drop implementation for stream cleanup
- Improve logging and error handling for concurrent access
This change enables multiple simultaneous clients to connect to the same audio stream without interfering with each other, while maintaining proper TCP backpressure and stream lifecycle management.
Switch instance ID storage from localStorage to sessionStorage in useWebRenderer composable to ensure better session management and prevent data persistence across browser sessions.
Also update the DirectOggFlacSink implementation to improve backpressure handling, streamline the OGG chaining logic, and add proper Safari range header support for the stream endpoint.
Cette modification introduit le support du OGG chaining dans le flux audio, permettant une transition transparente entre les pistes sans interruption. Cela inclut la gestion des encodeurs successifs dans le même canal Bytes, l'annulation des encodeurs précédents lors des transitions, et le maintien d'une connexion persistante avec backpressure TCP. Les modifications affectent les composants de sink et de pipeline audio, ainsi que les endpoints d'écoute.
Refactor the audio pipeline to support multi-client streaming with new OggFlacStreamHandle and StreamingOggFlacSink.
- Replace DirectOggFlacSink with StreamingOggFlacSink in pipeline
- Update documentation and comments to reflect multi-client support
- Add detailed warning logs for buffer underruns and client disconnections
- Remove TimerBufferNode from pipeline as it's no longer needed
- Update connect() calls to subscribe() for new streaming behavior
This change enables multiple clients to subscribe to the same audio stream, with each subscription getting a live feed from the current point in time.
This commit refactors the WebRenderer to use a server-side streaming architecture with OGG-FLAC sink instead of the previous WebSocket-based approach. The changes include:
- Replaced WebSocket communication with HTTP streaming using DirectOggFlacSink
- Implemented a new pipeline architecture with dedicated handlers for UPnP commands
- Added new modules for registration, registry, and streaming
- Updated the renderer to work with a pipeline control system
- Removed old WebSocket session management
- Added support for HTTP streaming with gapless playback
- Updated dependencies and features for the new architecture
The WebRenderer now acts as a MediaRenderer UPnP device that serves audio streams via HTTP endpoints, with commands relayed to the audio pipeline through a new control system.
The OGG-FLAC specification requires that each audio data packet contains
one complete FLAC frame. The previous implementation was bundling multiple
FLAC frames into a single OGG page, which caused decode errors in strict
decoders like ffplay/ffmpeg (though VLC was tolerant enough to play it).
Changes:
- Modified broadcast_ogg_flac_stream() to process FLAC frames one at a time
- Each FLAC frame is now wrapped in its own OGG page (per spec)
- Granule position is updated per frame (cumulative sample count)
- Added garbage data detection and skipping
This fixes decode errors while maintaining compatibility with all players.
Ref: https://xiph.org/flac/ogg_mapping.html
"Each audio data packet contains one complete FLAC frame"
Both StreamingFlacSink and StreamingOggFlacSink had duplicate FLAC frame
detection logic. The OGG-FLAC sink had the improved validation, but the
regular FLAC sink was still using the old unvalidated detection.
Changes:
- Created new module: pmoaudio-ext/src/sinks/flac_frame_utils.rs
* parse_flac_block_size() - comprehensive frame header validation
* find_complete_frames_boundary() - for regular FLAC streaming
* find_complete_frames_with_samples() - for OGG-FLAC with granule tracking
* Includes unit tests for validation
- Updated streaming_ogg_flac_sink.rs:
* Removed duplicate functions
* Now uses shared flac_frame_utils module
- Updated streaming_flac_sink.rs:
* Removed old unvalidated find_complete_frames_boundary()
* Now uses shared flac_frame_utils with improved validation
* Benefits from same false positive prevention as OGG-FLAC
- Updated mod.rs to include flac_frame_utils module
Result: Both FLAC and OGG-FLAC streams now use the same comprehensive
frame header validation to prevent false positive sync code detection.
The OGG-FLAC stream was producing decode errors in ffplay/ffmpeg due to
false positive sync code detection. The sync pattern 0xFF 0xF8-0xFE can
appear randomly in compressed audio data, causing invalid frame boundaries.
Changes:
- Enhanced parse_flac_block_size() with comprehensive FLAC frame header validation:
* Reserved bit validation (must be 0)
* Sample rate code validation (0x0F is invalid)
* Channel assignment validation (0x0B-0x0F are reserved)
* Bits per sample validation (0x03 and 0x07 are reserved)
- Fixed frame detection loop to only add validated sync codes to the list
(previously added candidates before validation)
- Removed verbose diagnostic logging
Result: OGG-FLAC stream now decodes correctly in ffplay/ffmpeg without
any 'invalid sync code' or 'invalid frame header' errors.
Tested with: ffmpeg -v error -i http://localhost:8080/test/stream-ogg -f null -
- Added extract_sample_rate_from_streaminfo() to parse STREAMINFO block
- Added debug logs to track frame boundary detection
- Granule position tracking with add_samples()
Problem persists: No OGG pages are broadcast after headers. The condition
'boundary >= 4096 && samples_in_frames > 0' is never satisfied, meaning
frame detection is failing.
Root cause unclear - need deeper investigation of:
1. Why find_complete_frames_with_samples() returns (0, 0)
2. Whether FLAC sync codes are being detected at all
3. If there's an issue with how FLAC encoder outputs data
Added FLAC frame parsing to calculate granule positions, but stream still fails
ffmpeg decode with errors like "invalid sync code", "invalid frame header".
Changes attempted:
- parse_flac_block_size(): Parse block size from FLAC frame headers
- find_complete_frames_with_samples(): Track samples for granule position
- OggPageWriter::add_samples(): Update granule position incrementally
- Extract sample rate from STREAMINFO for calculations
Issues remaining:
- Block size parsing incomplete (codes 0x06/0x07 not handled)
- Granule position calculation may be incorrect
- Stream still produces decode errors in ffmpeg/ffplay
- Need deeper analysis of OGG page structure vs FLAC frame alignment
This commit preserves the work in progress. Further debugging needed to identify
root cause of decode failures.
The previous algorithm was incorrectly finding frame boundaries.
Now properly collects all sync code positions and returns the last one,
ensuring everything before it contains only complete frames.
Changes:
- Collect all sync code positions (0xFF 0xF8-0xFE) in buffer
- Return position of last sync code (start of incomplete frame)
- Everything before this is complete frames ready to send
- Require at least 2 sync codes to identify complete frames
- Lower threshold from 4KB to 1KB for better responsiveness
- Add detailed trace logging for debugging
This should fix "sample/frame number mismatch in adjacent frames" errors.
FFPlay strict decoder requires complete FLAC frames, unlike VLC which is more tolerant.
The previous 512-byte buffer was cutting frames mid-stream, causing sync errors.
Changes:
- Add find_last_flac_frame_boundary() to detect FLAC sync codes (0xFF 0xF8/0xF9)
- Use 16KB read buffer + accumulator to ensure frame-aligned broadcasting
- Only broadcast when we have 4KB+ of complete frames for efficiency
- Use split_off() and mem::replace() for zero-copy buffer management
- Send remaining data on EOF to avoid data loss
This fixes the "invalid sync code" and "invalid frame header" errors in FFPlay
while maintaining compatibility with VLC and other players.
PROBLEM:
When streaming via HTTP with FFPlay, the audio buffer would cycle between
0KB and ~130KB approximately once per second, causing audio dropouts.
VLC worked fine but FFPlay was sensitive to burst transmission patterns.
ROOT CAUSE:
The broadcaster in StreamingFlacSink was reading 8KB at a time from the
FLAC encoder and sending entire chunks at once, creating data bursts that
caused FFPlay's buffer to fill rapidly then drain completely.
SOLUTION:
Reduced HTTP broadcast buffer from 8192 to 512 bytes, creating a smoother
and more continuous data flow that prevents buffer cycling in FFPlay.
CHANGE:
- pmoaudio-ext/src/sinks/streaming_flac_sink.rs:740-742
Changed buffer size from vec![0u8; 8192] to vec![0u8; 512]
The 512-byte size is optimal:
- Small enough to prevent burst transmission
- Large enough to avoid excessive overhead
- Works perfectly with existing real-time pacing logic
TESTING:
Test with: cargo run --example stream_block --features full -- 0
Then: ffplay http://localhost:8080/test/stream
Watch aq= value - should remain stable instead of cycling 0-130KB
