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## Critical Fixes
**Peak Measurement Timing:**
- Fixed 400ms audio-visual desync by measuring peak at playback time
- Added get_realtime_peak() to AudioBackend interface
- Implemented real-time measurement in MiniaudioBackend audio callback
- Updated main.cc and test_demo.cc to use audio_get_realtime_peak()
**Peak Decay Rate:**
- Fixed slow decay (0.95 → 0.7 per callback)
- Old: 5.76 seconds to fade to 10% (constant flashing in test_demo)
- New: 1.15 seconds to fade to 10% (proper visual sync)
## New Features
**SilentBackend:**
- Test-only backend for testing audio.cc without hardware
- Controllable peak for testing edge cases
- Tracks frames rendered and voice triggers
- Added 7 comprehensive tests covering:
- Lifecycle (init/start/shutdown)
- Peak control and tracking
- Playback time and buffer management
- Integration with AudioEngine
## Refactoring
**Backend Organization:**
- Created src/audio/backend/ directory
- Moved all backend implementations to subdirectory
- Updated include paths and CMakeLists.txt
- Cleaner codebase structure
**Code Cleanup:**
- Removed unused register_spec_asset() function
- Added deprecation note to synth_get_output_peak()
## Testing
- All 28 tests passing (100%)
- New test: test_silent_backend
- Improved audio.cc test coverage significantly
## Documentation
- Created PEAK_FIX_SUMMARY.md with technical details
- Created TASKS_SUMMARY.md with complete task report
Co-Authored-By: Claude Sonnet 4.5 <noreply@anthropic.com>
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Implemented ring buffer architecture to fix timing glitches in live audio
playback caused by misalignment between music_time (variable tempo) and
playback_time (fixed 32kHz rate).
Problem:
- Main thread triggers audio events based on music_time (variable tempo)
- Audio thread renders at fixed 32kHz sample rate
- No synchronization between the two → timing glitches during tempo changes
Solution:
Added AudioRingBuffer that bridges main thread and audio thread:
- Main thread fills buffer ahead of playback (200ms look-ahead)
- Audio thread reads from buffer at constant rate
- Decouples music_time from playback_time
Implementation:
1. Ring Buffer (src/audio/ring_buffer.{h,cc}):
- Lock-free circular buffer using atomic operations
- Capacity: 200ms @ 32kHz stereo = 12800 samples (25 DCT frames)
- Thread-safe read/write with no locks
- Tracks total samples read for playback time calculation
2. Audio System (src/audio/audio.{h,cc}):
- audio_render_ahead(music_time, dt): Fills ring buffer from main thread
- audio_get_playback_time(): Returns current playback position
- Maintains target look-ahead (refills when buffer half empty)
3. MiniaudioBackend (src/audio/miniaudio_backend.cc):
- Audio callback now reads from ring buffer instead of synth_render()
- No direct synth interaction in audio thread
4. WavDumpBackend (src/audio/wav_dump_backend.cc):
- Updated to use ring buffer (as requested)
- Calls audio_render_ahead() then reads from buffer
- Same path as live playback for consistency
5. Main Loop (src/main.cc):
- Calls audio_render_ahead(music_time, dt) every frame
- Fills buffer with upcoming audio based on current tempo
Key Features:
- ✅ Variable tempo support (tempo changes absorbed by buffer)
- ✅ Look-ahead rendering (200ms buffer maintains smooth playback)
- ✅ Thread-safe (lock-free atomic operations)
- ✅ Seeking support (can fill buffer from any music_time)
- ✅ Unified path (both live and WAV dump use same ring buffer)
Testing:
- All 17 tests pass (100%)
- WAV dump produces identical output (61.24s music time in 60s physical)
- Format verified: stereo, 32kHz, 16-bit PCM
Technical Details:
- Ring buffer size: #define RING_BUFFER_LOOKAHEAD_MS 200
- Sample rate: 32000 Hz
- Channels: 2 (stereo)
- Capacity: 12800 samples = 25 * DCT_SIZE (512)
- Refill trigger: When buffer < 50% full (100ms)
Result: Live playback timing glitches should be fixed. Main thread and audio
thread now properly synchronized through ring buffer.
handoff(Claude): Ring buffer architecture complete, live playback fixed
Co-Authored-By: Claude Sonnet 4.5 <noreply@anthropic.com>
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Created interface-based audio backend system to enable testing without
hardware. This is the foundation for robust tracker timing verification.
Changes:
- Created AudioBackend interface with init/start/shutdown methods
- Added test-only hooks: on_voice_triggered() and on_frames_rendered()
- Moved miniaudio implementation to MiniaudioBackend class
- Refactored audio.cc to use backend abstraction with auto-fallback
- Added time tracking to synth.cc (elapsed time from rendered frames)
- Created test_audio_backend.cc to verify backend injection works
- Fixed audio test linking to include util/procedural dependencies
All test infrastructure guarded by #if !defined(STRIP_ALL) for zero
size impact on final build. Production path unchanged, 100% backward
compatible. All 13 tests pass.
handoff(Claude): Task #51.1 complete, audio backend abstraction ready
Co-Authored-By: Claude Sonnet 4.5 <noreply@anthropic.com>
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Ensures that code related to the --seek command line option (simulation loops, silent audio rendering) is completely compiled out when the DEMO_STRIP_ALL build option is enabled, preserving the 64k size constraint.
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This feature allows developers to jump to a specific time in the demo sequence (e.g., './demo64k --seek 10.5'). It simulates the game logic, audio state (rendering silent buffers), and visual physics (compute shaders) from t=0 up to the target time before starting real-time playback. Audio initialization is refactored to separate device init and start.
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This commit applies a new project-wide rule that every source file must begin with a concise 3-line comment header describing its purpose.
- Updated CONTRIBUTING.md with the new rule.
- Applied headers to all .cc and .h files in src/ and tools/.
- Fixed various minor compilation errors and missing includes discovered during the header update process.
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