Problem:
When switching audio sources (USB to HDMI or vice versa), the application
would hang indefinitely. This was caused by a deadlock between Go and C
layers:
1. Main thread calls SetAudioOutputSource() → stopOutputAudio()
2. stopOutputAudio() calls outputRelay.Stop() which waits for goroutine
3. Goroutine is blocked in ReadMessage() holding Go mutex
4. ReadMessage() calls blocking C function jetkvm_audio_read_encode()
5. C function is blocked reading from ALSA device
6. Disconnect() can't acquire Go mutex to clean up
7. Deadlock: Main thread waiting for goroutine, goroutine waiting for ALSA
Solution:
Release the Go mutex BEFORE calling blocking C functions in ReadMessage()
and WriteMessage(). The C layer has its own pthread mutex protection and
handles stop requests via atomic flags. This allows:
- Disconnect() to acquire the mutex immediately
- C layer to detect stop request and return quickly
- Goroutines to exit cleanly
- Audio source switching to work flawlessly
Fixes:
- internal/audio/cgo_source.go:ReadMessage() - Release mutex before C call
- internal/audio/cgo_source.go:WriteMessage() - Release mutex before C call
This fix eliminates the hang when switching between USB and HDMI audio
sources.
Root cause: ALSA assertion failure `snd_pcm_writei: Assertion 'pcm' failed`
when pcm_playback_handle became NULL during concurrent write operations.
The crash occurred because:
1. Thread A checks pcm_playback_handle != NULL (passes)
2. Thread B calls jetkvm_audio_playback_close(), sets handle = NULL
3. Thread A calls snd_pcm_writei(NULL, ...) → SIGABRT
Solution: Added pthread mutexes to protect concurrent access:
- playback_mutex protects pcm_playback_handle in decode_write and close
- capture_mutex protects pcm_capture_handle in read_encode and close
All critical sections now acquire mutex before accessing ALSA handles,
preventing the NULL pointer from being passed to ALSA functions.
Root cause: ALSA was silently using a different sample rate than configured,
causing severe pitch/speed distortion (the "cassette player" warping effect).
The bug occurred when:
- User configured 48 kHz in UI
- HDMI source output 44.1 kHz audio
- set_rate() failed, set_rate_near() chose 44.1 kHz
- Code never checked what rate was actually set
- Opus encoder created for 48 kHz but received 44.1 kHz audio
- Result: ~9% pitch shift and timing mismatch
Fix:
- Always use set_rate_near() and check the actual rate returned
- Pass detected rate and frame size to Opus encoder/decoder creation
- Avoid modifying global state to prevent capture/playback interference
- Recalculate frame_size for 20ms at the actual rate
- Verify rate after hw_params application
- Add detailed logging for rate adaptation
This ensures Opus encoder/decoder use the correct rate matching the hardware,
regardless of what the HDMI source outputs.
- Update default EDID with registered manufacturer ID (Dell) and proper 24-inch display dimensions (52x32cm) for better macOS/OS compatibility
- Add configurable sample rate (32/44.1/48/96 kHz) to support different HDMI audio sources
- Add packet loss compensation percentage control for FEC overhead tuning
- Fix config migration to ensure new audio parameters get defaults for existing configs
- Update all language translations for new audio settings
- Add config fields for bitrate, complexity, DTX, FEC, buffer periods
- Add RPC methods for get/set audio config and restart
- Add UI settings page with controls for all audio parameters
- Add Apply Settings button to restart audio with new config
- Add config migration for backwards compatibility
- Add translations for all 9 languages
- Clean up redundant comments and optimize log levels
- Revert .golangci.yml to dev branch state (removed custom build-tags)
- Remove internal/audio/cgo_source_stub.go (not needed with proper cross-compilation)
- Fix import ordering in ui/src/utils.ts
Use 'make lint-go' for proper ARM cross-compilation environment.
Provides no-op AudioSource implementations for platforms that don't
support ARM CGO audio (x86_64, darwin, etc.). This allows golangci-lint
to run successfully on any platform without requiring ARM cross-compilation
toolchain.
The stub implementations return errors when called, ensuring that if
they're accidentally used at runtime on non-ARM platforms, it will fail
gracefully with a clear error message rather than undefined symbols.
Build constraints ensure the real CGO implementation is used on linux/arm
and linux/arm64, while stubs are used everywhere else.
Remove all subprocess-based audio code to simplify the audio system and
reduce complexity. Audio now uses CGO in-process mode exclusively.
Changes:
- Remove subprocess mode: Deleted Supervisor, IPCSource, embed.go
- Remove audio mode selection from UI (Settings → Audio)
- Remove audio mode from backend config (AudioMode field)
- Remove JSON-RPC handlers: getAudioMode/setAudioMode
- Remove Makefile targets: build_audio_output/input/binaries
- Remove standalone C binaries: jetkvm_audio_{input,output}.c
- Remove IPC protocol implementation: ipc_protocol.{c,h}
- Remove unused IPC functions from audio_common.{c,h}
- Simplify audio.go: startAudio() instead of startAudioSubprocesses()
- Update all function calls and comments to remove subprocess references
- Add constants to cgo_source.go (ipcMaxFrameSize, ipcMsgTypeOpus)
- Keep update_opus_encoder_params() for potential future runtime config
Benefits:
- Simpler codebase: -1,734 lines of code
- Better performance: No IPC overhead on embedded hardware
- Easier maintenance: Single audio implementation
- Smaller binary: No embedded audio subprocess binaries
The audio system now works exclusively via CGO direct C function calls,
with ALSA device selection (HDMI vs USB) still configurable via settings.
Audio quality improvements:
- Enable constrained VBR to prevent bitrate starvation at low volumes
- Increase Opus complexity from 2 to 5 for better quality
- Enable DTX for bandwidth optimization
- Enable FEC (Forward Error Correction)
- Add DTX and FEC signaling in SDP (usedtx=1;useinbandfec=1)
Default configuration changes:
- Change default audio output source from HDMI to USB
- Enable USB Audio device by default
- USB audio works on current stable image (HDMI requires newer device tree)
These changes fix crackling issues at low volumes and provide better
overall audio quality for both USB and HDMI audio paths.
Remove dynamic gain code and rely on Opus encoder quality improvements:
- Increase Opus complexity from 2 to 5 for better quality
- Change bandwidth from FULLBAND (20kHz) to SUPERWIDEBAND (16kHz) for better quality at 128kbps
- Disable FEC to allocate all bits to audio quality
- Increase ALSA buffer from 40ms to 80ms for stability
The dynamic gain code was adding complexity without solving the underlying
issue: TC358743 HDMI chip captures digital audio at whatever volume the
source outputs. Users should adjust volume at the source or in their browser.
Add noise gate threshold at peak > 256 (-42dB) to prevent dynamic gain
from amplifying quantization noise and hardware noise floor. This fixes
crackling, buzzing, and static-like noise when HDMI audio is at very
low volume or during silence.
Without the gate, signals below -42dB (peak < 256) would get 8x gain
applied, amplifying noise floor to audible levels. Now these signals
pass through unmodified, eliminating the artifacts.
- Check SetReadDeadline error in IPC client
- Explicitly ignore Kill() error (process may be dead)
- Remove init() function and rely on explicit ExtractEmbeddedBinaries() call
Implement SIMD-optimized audio operations using ARM NEON for Cortex-A7 targets
Update Makefile and CI configuration to support NEON compilation flags
Add SIMD implementations for common audio operations including:
- Sample clearing and interleaving
- Volume scaling and format conversion
- Channel manipulation and balance adjustment
- Endianness swapping and prefetching
Alex P