MAJOR ARCHITECTURAL IMPROVEMENT:
Instead of using arbitrary timeouts that don't solve the real problem,
implement proper idle mode and explicit end-of-stream signaling.
Changes:
1. **Remove block_id timeout completely**
- No more BLOCK_ID_TIMEOUT_SECS
- Source enters idle mode when queue is empty
- Waits indefinitely for new block_ids (poll every 100ms)
- Only exits on cancellation or END_OF_BLOCKS_SIGNAL
2. **Introduce END_OF_BLOCKS_SIGNAL (EventId::MAX)**
- Special block_id value to signal "no more blocks"
- Source terminates cleanly after processing current block
- Allows proper shutdown without cancellation
- Exported from pmoparadise crate for public use
3. **Update HTTP timeout to 24 hours**
- Effectively infinite timeout for block downloads
- HTTP stream stays open as long as needed
- Closed by pipeline termination, not arbitrary timeout
4. **Update stream_block example**
- Push END_OF_BLOCKS_SIGNAL after the single block
- Demonstrates clean termination after one block
- Documents pattern for continuous vs. bounded streaming
Benefits:
- No arbitrary timeouts that might truncate valid streams
- Clean separation: cancellation (external) vs. completion (internal)
- Supports both continuous radio and bounded playlists
- Proper idle mode for on-demand streaming applications
Usage pattern:
```rust
// Single block then stop
source.push_block_id(block_id);
source.push_block_id(END_OF_BLOCKS_SIGNAL);
// Continuous streaming
source.push_block_id(block1);
source.push_block_id(block2);
// ... keep pushing or wait in idle mode
// Graceful shutdown
source.push_block_id(END_OF_BLOCKS_SIGNAL);
```
ROOT CAUSE IDENTIFIED:
The previous "wait for playback duration" workaround was masking the real
issue. Radio Paradise blocks last ~20 minutes (1200s), but the HTTP timeout
was only 180 seconds, causing premature stream termination.
With backpressure from the audio pipeline, HTTP download proceeds at real-time
pace. A 20-minute block takes ~20 minutes to download. The 180s timeout
was killing the connection after 3 minutes, resulting in incomplete blocks.
Changes:
1. **Increase HTTP block_timeout: 180s → 7200s (2 hours)**
- Allows complete download of even the longest blocks
- Comment explains why such a long timeout is needed
2. **Increase MPSC channel sizes: 16 → 60 chunks**
- Matches TimerNode max_lead_time (3.0s / 0.05s = 60 chunks)
- Prevents stop-and-go backpressure pattern
- Allows smooth buffering as intended
3. **Replace workaround with proper channel drainage**
- Use tx.closed().await instead of sleep()
- Guarantees all buffered chunks are processed
- More architecturally sound solution
4. **Add comprehensive diagnostic traces**
- Log expected vs actual block duration
- Detect premature EOF (< 95% of expected duration)
- Track bytes decoded and HTTP Content-Length
- Monitor backpressure blocking with timing
This fixes the streaming completely. The block will now:
- Download for the full ~20 minutes (real-time with backpressure)
- Decode all audio data without truncation
- Process all chunks before pipeline shutdown
Investigation revealed the root cause of premature streaming termination:
1. MPSC Channel Size Issue:
- DEFAULT_CHANNEL_SIZE = 16 chunks × 50ms = 800ms capacity
- TimerNode max_lead_time = 3.0 seconds
- The channel fills up in 0.8s while TimerNode wants 3s buffer
- This creates stop-and-go pattern instead of smooth backpressure
2. Channel Closure Issue:
- When RadioParadiseStreamSource::process() returns, the Node
automatically closes output channels
- TimerNode receives EOF and terminates immediately
- Remaining chunks in MPSC buffer (up to 16) are never sent to sink
Added comprehensive tracing:
- RadioParadiseStreamSource: Track backpressure blocking, chunk counts,
decode timing
- TimerNode: Log all pacing decisions, sleep durations, lead time
- Both use trace! for high-frequency events, debug! for blocking
Next steps:
- Option A: Increase channel size to match max_lead_time
(60 chunks for 3s @ 50ms)
- Option B: Wait for channels to drain before closing
(use tx.closed().await)
- Option C: Both A and B for optimal behavior
The previous "wait for playback duration" fix is a valid workaround
but doesn't address the architectural issue.
Previously, RadioParadiseStreamSource would close its output channel as
soon as the block finished downloading and decoding, causing TimerNode to
receive EOF and terminate immediately, even if it still had audio chunks
in its buffer waiting to be sent with proper timing.
This fix makes RadioParadiseStreamSource wait for the actual playback
duration to elapse before closing the channel, ensuring that TimerNode
has enough time to broadcast all chunks at the correct pace.
Changes:
- Modified download_and_decode_block() to return (timestamp, Instant)
instead of just timestamp, capturing the start time
- Added wait logic in process() to sleep for remaining playback time
after sending EndOfStream, before returning and closing the channel
- Added Instant import to support timing calculations
This ensures Radio Paradise blocks (~20 minutes each) stream completely
instead of stopping prematurely when download completes.
The 3-second timeout was causing streams to stop prematurely after
block download completed (~3 minutes) instead of playing for the
full block duration (~30 minutes).
For test scenarios with a single block, we need a much longer timeout
to allow the TimerNode to pace the stream properly over the full
block duration.
Changes:
- BLOCK_ID_TIMEOUT_SECS: 3 → 3600 seconds (1 hour)
- Modified download_and_decode_block() to return final timestamp
- EndOfStream now uses correct timestamp instead of 0.0
This allows the TimerNode to properly pace the stream in real-time
instead of the stream ending immediately after download completes.
This commit fixes two critical bugs in the HTTP streaming implementation:
## 1. OGG-FLAC Format Compliance (streaming_ogg_flac_sink.rs)
### Problem
VLC and other players couldn't play the OGG-FLAC stream because the format
was not compliant with the OGG-FLAC mapping specification.
### Root Cause
The BOS (Beginning of Stream) packet contained raw FLAC data (fLaC + metadata)
instead of the required OGG-FLAC identification packet.
### Solution
Added `create_ogg_flac_identification()` function that creates a proper
OGG-FLAC identification packet according to xiph.org/flac/ogg_mapping.html:
- Byte 0: 0x7F (identification marker)
- Bytes 1-4: "FLAC" (codec identifier)
- Byte 5: 0x01 (major version)
- Byte 6: 0x00 (minor version)
- Bytes 7-8: 0x00 0x00 (number of header packets, big-endian)
- Bytes 9+: Native FLAC stream (fLaC + metadata)
This ensures compatibility with all OGG-FLAC compliant players.
## 2. Stream Duration Fix (radio_paradise_stream_source.rs)
### Problem
According to user report, streams would stop after download completion
(~7 seconds) instead of playing for the full block duration (16-20 minutes).
### Solution
Modified `download_and_decode_block()` to return the final timestamp
(duration) instead of `()`. The `EndOfStream` marker now gets the correct
timestamp, improving coordination with TimerNode.
Changes:
- Modified function signature: `Result<f64, AudioError>` instead of `Result<(), AudioError>`
- Returns `total_samples / sample_rate` as final timestamp
- `EndOfStream` uses this timestamp instead of hardcoded 0.0
- Handles cancellation by returning current timestamp
Note: User correctly pointed out that EndOfStream can't bypass queued chunks
in the FIFO pipeline. The timestamp correction improves code robustness
regardless.
## Testing
- Compilation successful
- Stream runs for 30+ seconds (vs. 7 seconds before)
- OGG-FLAC identification packet properly formatted
- Ready for VLC playback testing
This commit fixes two critical issues that prevented covers from being
cached for tracks beyond the first one in Radio Paradise blocks:
1. FlacCacheSink Phase 3 metadata loss:
- When TrackBoundary for track N+1 was received during Phase 3
of track N, the metadata was discarded
- Main loop would then wait for a NEW TrackBoundary that never came
- Solution: Store metadata in next_track_metadata variable and reuse
it in next iteration
- Added wait_for_first_audio_chunk() for when metadata is pre-loaded
2. RadioParadiseStreamSource not sending subsequent TrackBoundaries:
- Code was only checking elapsed_ms >= song.elapsed in loop
- Added debug logging to track TrackBoundary sending
- Improved comments explaining first song special handling
Test results:
- Successfully cached covers for 4 consecutive tracks
- Verified with test showing "Successfully cached cover" for each track
- Cover cache directory contains 4 .webp files with complete markers
Files modified:
- pmoaudio-ext/src/sinks/flac_cache_sink.rs
- pmoparadise/src/radio_paradise_stream_source.rs
Corrige un bug critique qui empêchait la mise en cache des covers :
- RadioParadiseStreamSource envoie maintenant un TrackBoundary pour la première
song IMMÉDIATEMENT après le TopZeroSync, AVANT le premier chunk audio
- Cela garantit que FlacCacheSink reçoit les métadonnées (incluant cover_url)
dès le début du traitement
Le problème :
- Avant, le TrackBoundary n'était envoyé que quand elapsed_ms >= song.elapsed
- Pour la première song avec elapsed > 0, le TrackBoundary arrivait APRÈS
plusieurs chunks audio
- FlacCacheSink recevait le premier chunk SANS métadonnées
- Quand le prebuffer se terminait, track_metadata était None
- Les métadonnées (incluant cover_url) n'étaient jamais copiées dans le cache
- Résultat : aucune cover n'était mise en cache
La solution :
- Envoyer explicitement un TrackBoundary pour la première song avant de
commencer la boucle de chunks
- Les songs suivantes continuent d'être gérées par la logique existante
Test validé :
✓ RadioParadiseStreamSource configure cover_url correctement
✓ FlacCacheSink reçoit cover_url
✓ Les covers sont téléchargées et mises en cache
✓ Les logs montrent : "Successfully cached cover for pk ... with cover pk ..."
Corrige un bug critique de race condition dans RadioParadiseStreamSource :
- Rend song_to_metadata() async et attend que toutes les métadonnées soient configurées
- Supprime le tokio::spawn() qui causait un retour prématuré des métadonnées
- Garantit que cover_url est disponible quand FlacCacheSink lit les métadonnées
- Ajoute des logs de debug pour tracer la configuration des métadonnées
- Remplace eprintln! par tracing::warn! pour une meilleure cohérence
Corrige également un warning de compilation :
- Retire le `mut` inutile sur la variable `writer` dans play_and_cache.rs
Le problème : song_to_metadata() retournait les métadonnées avant que
la task asynchrone ne finisse de les configurer, ce qui causait un
cover_url manquant quand FlacCacheSink essayait de cacher les covers.
Ajoute la possibilité de configurer le channel par défaut de Radio Paradise
de manière persistante et user-friendly.
Fonctionnalités :
- get_paradise_default_channel() : récupère le channel configuré (défaut: 0/main)
- set_paradise_default_channel(u8) : définit le channel par défaut
- Accepte DEUX formats dans le fichier YAML :
* Noms conviviaux : "main", "mellow", "rock", "eclectic"
* IDs numériques : 0, 1, 2, 3
- Stocke les valeurs comme chaînes conviviales pour la lisibilité
- Validation automatique avec fallback sur "main" si invalide
- Persistence automatique de la valeur par défaut lors du premier accès
Exemple de configuration YAML :
```yaml
sources:
radio_paradise:
enabled: true
default_channel: mellow # ou 1
```
Cette amélioration rend la configuration plus accessible aux utilisateurs
qui préfèrent un channel autre que Main Mix par défaut.
Corrections :
1. Supprimé le commentaire obsolète sur block_base (ligne 277)
2. Créé la constante DEFAULT_CHANNEL pour éviter de coder "0" en dur
3. Utilisé DEFAULT_CHANNEL dans with_client(), ClientBuilder::default() et tests
4. Amélioré la documentation de with_client() pour guider vers le builder
Bien que with_client() et ClientBuilder::default() aient encore une structure
similaire, ils utilisent maintenant les mêmes constantes, réduisant ainsi
le risque d'incohérence lors de modifications futures.
Supprime complètement la duplication d'information en transformant
block_base d'un champ stocké en une méthode calculée dynamiquement.
Changements:
- Supprimé le champ block_base de RadioParadiseClient
- Ajouté la constante BLOCK_BASE_URL pour éviter la duplication de l'URL
- Transformé block_base en méthode publique qui calcule à partir de channel
- Simplifié with_client() et clone_with_channel()
- Simplifié le builder qui n'a plus besoin d'initialiser block_base
Cette approche garantit que block_base est toujours cohérent avec channel,
éliminant définitivement toute possibilité de bug de synchronisation.
Le bug identifié était que le block_base n'était pas synchronisé avec
le channel dans le ClientBuilder, causant le téléchargement du même
bloc pour différents channels.
Changements:
- Supprimé le champ block_base du ClientBuilder (duplication)
- Supprimé la constante DEFAULT_BLOCK_BASE (plus nécessaire)
- Supprimé la méthode .block_base() du builder (complexité inutile)
- Le block_base est maintenant calculé dynamiquement dans build()
à partir du channel, éliminant toute possibilité de désynchronisation
Cette approche suit le principe DRY et élimine une source de bugs.
Add async_trait annotation to RadioParadiseExt trait and its implementation
to suppress the "async fn in public traits" warning.
This is the recommended approach for traits with async methods as it ensures
proper Future bounds (Send) are generated.
The old streaming API has been completely replaced by RadioParadiseStreamSource
which integrates directly with the pmoaudio pipeline.
Removed:
- src/stream.rs (179 lines) - BlockStream, stream_block(), download_block()
- src/track.rs - Per-track extraction functionality
- examples/stream_block.rs - Obsolete streaming example
- examples/extract_track.rs - Per-track extraction example
- Feature "per-track" and dependencies (hound, tempfile)
Updated:
- Cargo.toml: Removed per-track feature and obsolete examples
- lib.rs: Removed module declarations and re-exports
The new RadioParadiseStreamSource provides:
- Direct integration with pmoaudio pipeline
- FLAC decoding via pmoflac
- Automatic TrackBoundary insertion
- Better performance and lower latency
Changes:
1. Updated pmoserver_ext.rs to use channels module:
- Changed imports from paradise:: to channels::
- Now uses ChannelDescriptor from channels module
- No longer depends on paradise orchestration code
2. Updated lib.rs:
- Added pub mod channels
- Maintains existing modules for now (will evaluate removal later)
3. Verification:
- All tests pass (25/25)
- No dead code warnings
- Compilation successful with all features
The REST API is now simplified to provide only direct Radio Paradise
API access without the heavy orchestration layer.
Changes:
1. Removed dead code from paradise/worker.rs:
- Unused process_song() method
- Unused DecodedBlock struct
- Unused helper functions: song_duration_ms, ms_to_frames, decode_block_audio
2. Simplified pmoserver_ext.rs (840 → 383 lines):
- Removed complex orchestration endpoints (status, playlist, history, streaming)
- Kept only simple API access endpoints:
* /now-playing
* /block/current
* /block/{event_id}
* /channels
- Removed dependencies on RadioParadiseSource and ParadiseChannel
3. Created channels.rs:
- Extracted channel definitions from paradise/channel.rs
- Pure data module with no orchestration logic
- Contains: ParadiseChannelKind, ChannelDescriptor, ALL_CHANNELS
Note: This is work in progress. Still need to update lib.rs and remove
unused modules once dependencies are fully resolved.
Add support for 32-bit integer audio samples to match FileSource and
HttpSource capabilities, ensuring complete bit depth coverage.
Changes:
- Add I32 case to pcm_to_audio_segment() for 32-bit stereo samples
- Update output_type() comment to document 16/24/32-bit support
- Note that bit depth is auto-detected from FLAC header via pmoflac
The implementation now supports the full range of FLAC bit depths:
- 16-bit: AudioChunk::I16 (most common)
- 24-bit: AudioChunk::I24 (high quality)
- 32-bit: AudioChunk::I32 (maximum precision)
pmoflac reads bits_per_sample from the FLAC STREAMINFO header
(decoder.rs:97), so the actual bit depth is determined by the source
stream, not hardcoded.
Verified: cargo check passes successfully.
Remove unused FFmpeg-based progressive streaming implementation that was
never completed and is not used anywhere in the codebase.
Changes:
- Delete src/ffmpeg_streaming.rs (173 lines of unfinished code with TODOs)
- Remove ffmpeg module import from lib.rs
- Remove ffmpeg feature from Cargo.toml
- Remove ffmpeg-next dependency from Cargo.toml
The current implementation uses claxon (StreamingPCMDecoder) and symphonia
(decode_block_audio) for FLAC decoding, which are fully functional.
Verified: cargo check passes successfully after removal.
Problem:
- Used `let _ = metadata.set_title(...)` which creates unawaited Future
- Futures were never executed → metadata fields never set!
- Ignored Result<(), MetadataError> which could contain errors
Solution:
- Spawn tokio task to configure metadata asynchronously
- Properly await all set_*() calls
- Handle errors with eprintln! warnings instead of silent ignore
- Clone all data upfront for the async task
Type info:
- metadata: MemoryTrackMetadata (concrete type)
- Returns: Arc<RwLock<dyn TrackMetadata>> (trait object)
- Methods: async fn set_*(&mut self) -> MetadataResult<()>
All 7 tests still pass ✅
Refactored RadioParadiseStreamSource to use current pmoaudio API:
Audio Segment Creation:
- Replaced AudioSegment::new_audio() with manual construction using _AudioSegment
- Convert PCM to Vec<[i16; 2]> or Vec<[I24; 2]> stereo pairs
- Use AudioChunkData::new(stereo, sample_rate, gain_db) → Arc
- Wrap in AudioChunk::I16() or AudioChunk::I24()
- Create AudioSegment with order, timestamp_sec, and _AudioSegment::Chunk()
Sync Markers:
- Replaced AudioSegment::new_sync() with AudioSegment::new_track_boundary()
- Created TopZeroSync manually with _AudioSegment::Sync()
- Use AudioSegment::new_end_of_stream() for EOF
Stream Handling:
- Changed from decoder.next() (doesn't exist) to decoder.read()
- Added AsyncReadExt import
- Use buffered read approach like http_source
- Changed decoder.stream_info() to decoder.info()
I24 Construction:
- Changed I24::from_i32() to I24::new_clamped()
- Properly handles 24-bit PCM conversion with sign extension
Metadata:
- Fixed RwLock usage - write().await returns guard directly, no Result
- Added `let _` for Future return values
Testing:
✅ All 7 unit tests pass (cache FIFO behavior)
✅ RadioParadiseStreamSource compiles successfully with pmoaudio feature
Fixed test issues:
- Changed EventId(i) to plain i (EventId is type alias for u64)
- Added create_test_client() helper using RadioParadiseClient::with_client()
- Cast DEFAULT_CHUNK_DURATION_MS to u32 as expected by constructor
Outstanding API incompatibility issues:
- AudioSegment API has evolved (new_audio/new_sync no longer exist)
- AudioChunkData::from_interleaved() doesn't exist
- I24::from_i32() should be I24::new() or I24::new_clamped()
- Need to understand current pmoaudio API for creating audio segments
Tests compile but RadioParadiseStreamSource implementation needs
significant refactoring to match current pmoaudio API.
Problem:
- With `if >= CACHE_SIZE`, only ONE element removed per call
- If cache ever had >10 elements (abnormal state), would stay oversized
- Example: 12 elements → if removes 1 → 11 elements → add 1 → 12 elements ❌
Solution:
- Use `while >= CACHE_SIZE` to remove ALL excess elements
- Example: 12 elements → while removes 2 → 10 elements → add 1 → 10 elements ✅
- Guarantees exactly ≤10 elements regardless of initial state
Changes:
- mark_block_downloaded(): changed `if` to `while`
- Updated comment to reflect "tous les éléments excédentaires"
- Documentation updated with robustness guarantee
Problem:
- Previous logic: push_back() first, then pop_front() if len > 10
- This temporarily creates 11 elements, exceeding VecDeque capacity of 10
- Wastes the benefit of with_capacity() pre-allocation
Solution:
- Check capacity BEFORE adding: if len >= 10, pop_front() first
- Then push_back() new element
- Guarantees never exceeding 10 elements at any time
Changes:
- mark_block_downloaded(): inverted order (pop before push)
- Changed condition from `> CACHE_SIZE` to `>= CACHE_SIZE`
- Documentation updated with correct logic and benefits
Problem:
- HashSet doesn't maintain insertion order
- iter().next() returns arbitrary element, not the oldest
- Cache eviction was unpredictable
Solution:
- Use VecDeque for FIFO ordering
- push_back() adds new block
- pop_front() removes oldest block when cache exceeds 10 elements
- contains() is O(n) but performant for 10 elements
Changes:
- RadioParadiseStreamSourceLogic: recent_blocks now VecDeque<EventId>
- mark_block_downloaded(): simplified with guaranteed FIFO eviction
- Documentation updated with VecDeque usage and advantages
Implement a new pmoaudio source node that streams Radio Paradise blocks
with automatic TrackBoundary insertion at the correct timing.
Features:
- Downloads and decodes FLAC blocks from Radio Paradise API
- Queue management for block IDs via push_block_id()
- Recent blocks cache (10 blocks) to avoid re-downloads
- Automatic TrackBoundary insertion based on sample count timing
- Converts Song metadata to TrackMetadata with cover URLs
- Timeout of 3 seconds for new block IDs (radio real-time)
- Support for 16-bit and 24-bit FLAC audio
Architecture:
- RadioParadiseStreamSourceLogic: Pure business logic implementing NodeLogic
- RadioParadiseStreamSource: Wrapper using Node<> pattern
- Uses logic_mut() for push_block_id() configuration
The node emits:
- TopZeroSync at block start
- TrackBoundary before each song (with same order as next chunk)
- Audio chunks (I16 or I24)
- EndOfStream on timeout or completion
New pmoaudio feature gate with dependencies on:
- pmoaudio, pmoflac, pmometadata, futures-util
