- Introduce `Streamtype` enum (Continuous vs Finite) to distinguish radio streams from finite tracks
- Enrich `TrackBoundary` sync marker with stream type for proper pause behavior per mode (silence vs backpressure)
- Update all sources and sinks to pass `StreamType` when creating track boundaries
- Radio Paradise, HTTP source → Continuous (infinite)
- Improve UPnP control architecture: pause sends silence for radio, blocks pipeline via backpressure for tracks
- Prepare groundwork for multi-client DSP architecture with shared source and per-DSP pipelines
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 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 Radio Paradise blocks via HTTP using FFPlay, the buffer
would cycle between 0KB and ~130KB approximately once per second, causing
audio dropouts and interruptions. VLC worked fine, but FFPlay was sensitive
to the burst transmission pattern.
ROOT CAUSE:
In StreamingFlacSink::broadcast_flac_stream(), the broadcaster was reading
8KB (8192 bytes) at a time from the FLAC encoder and sending the entire
chunk at once to all HTTP clients. This created a "burst" pattern:
- Read 8KB from encoder
- Send entire 8KB chunk to all clients
- Sleep if ahead of real-time pacing
- Repeat
FFPlay's buffer would fill rapidly with each 8KB burst, then drain completely
before the next burst arrived, causing the observed cycling behavior.
SOLUTION:
Reduced the HTTP broadcast buffer size from 8KB to 512 bytes in
StreamingFlacSink::broadcast_flac_stream(). This creates a much smoother,
more continuous data flow that FFPlay can handle without buffer cycling.
The 512-byte buffer size is:
- Small enough to prevent burst transmission
- Large enough to avoid excessive overhead
- Sufficient for smooth streaming with real-time pacing
CHANGES:
- Restore stream_block.rs example from git history
- Restore StreamingFlacSink and StreamingOggFlacSink from git history
- Add "streaming" feature to pmoaudio-ext
- Reduce broadcast buffer from 8192 to 512 bytes
- Update pmoparadise to use pmoaudio-ext streaming feature
TESTING:
Test with FFPlay to verify smooth buffering:
```bash
cargo run --example stream_block --features full -- 0
# In another terminal:
ffplay http://localhost:8080/test/stream
```
Watch the "aq=" value in FFPlay output. It should now remain stable
instead of cycling between 0KB and 130KB.
