ALSA now forces the configured sample rate (default 48kHz) instead of
auto-detecting the source rate. This prevents Opus encoder initialization
failures when HDMI sources output 44.1kHz audio, which Opus doesn't support.
Changes:
- Use snd_pcm_hw_params_set_rate() to force exact rate (48kHz by default)
- ALSA performs software resampling if hardware rate differs
- Update valid rates to Opus-compatible only (8k, 12k, 16k, 24k, 48k)
- Remove auto-adaptation logic that caused Opus failures with 44.1kHz
This ensures audio capture works reliably with any HDMI source rate.
- Fix validateAndApply comment to clarify it returns values, doesn't apply them
- Correct capture_channels comment about hardware capabilities
- Fix opus_packet_loss_perc default value from 0 to 20 (matches backend default)
- Fix handle_alsa_error return value documentation (return 0 also unlocks mutex)
Changes:
- Consolidate duplicate stop logic into helper functions
- Fix RPC getAudioConfig to return actual runtime values instead of
inconsistent defaults (bitrate was returning 128 vs actual 192)
- Improve setAudioTrack mutex handling to eliminate nested locking
- Simplify ALSA error retry logic by reorganizing conditional branches
- Split CGO Connect() into separate input/output methods for clarity
- Use map lookup for sample rate validation instead of long if-chain
- Add inline comments documenting validation steps
All changes preserve existing functionality while reducing code
duplication and improving readability. Tested with both HDMI and
USB audio sources.
Critical Fixes:
- Fix race condition in handleInputTrackForSession by reloading source inside mutex
- Fix ALSA handle cleanup atomicity (nullify before close to prevent use-after-free)
- Bounds check for opus buffer already present (verified)
Configuration Alignment:
- Align audio bitrate default to 192 kbps across all layers (C, Go defaults, config)
- Align audio complexity default to 8 across all layers
- Align DTX default to enabled (true/1) across all layers for bandwidth efficiency
Documentation Improvements:
- Update C header comment to reflect accurate 192 kbps default
- Clarify NEON requirement (not just "always available")
- Fix ALSA device mapping comments to reflect environment variable usage
- Document fallback behavior in playback init
Code Quality:
- Add validation logging for out-of-range audio configuration values
- Improve error visibility for configuration issues
All changes thoroughly analyzed before implementation.
Change recovery_attempts from int to uint8_t for better efficiency:
- Reduces memory footprint (1 byte vs 4 bytes)
- Better cache utilization on ARM
- Matches max_attempts type (uint8_t)
- Values never exceed 3, fits perfectly in uint8_t range
Updated function signature and all call sites for consistency.
Extract shared error recovery logic:
- Create handle_alsa_error() for EPIPE, EAGAIN, ESTRPIPE, EIO errors
- Consolidates ~180 lines of duplicate error handling code
- Used by both capture and playback paths
Extract shared close logic:
- Create close_audio_stream() for safe shutdown sequence
- Handles CAS synchronization, delay, mutex protection
- Used by both jetkvm_audio_capture_close and jetkvm_audio_playback_close
Remove all TRACE_LOG dead code:
- TRACE_LOG was compiled to ((void)0) with zero runtime value
- Eliminates ~30 statements cluttering the codebase
Result: 87 lines removed (9% reduction), improved maintainability
- Separate capture_channels (stereo HDMI) from playback_channels (mono mic)
to prevent initialization conflicts that were breaking stereo output
- Optimize defaults for LAN use: 192kbps bitrate, complexity 8, 0% packet
loss compensation, DTX disabled (eliminates static and improves clarity)
- Add comprehensive race condition protection in C audio layer with handle
validity checks and mutex-protected cleanup operations
- Enable USB audio volume control and configure microphone as mono
- Add centralized AUDIO_DEFAULTS constant in UI with localized labels
- Add missing time import to fix compilation
This resolves audio quality issues and crash scenarios when switching
between HDMI and USB audio sources.
Problem:
Previous fix reduced but didn't eliminate the hang when switching audio
sources. The C layer was still blocking on snd_pcm_readi()/snd_pcm_writei()
while holding the mutex, preventing cleanup from proceeding.
Solution:
Call snd_pcm_drop() BEFORE acquiring the mutex in close functions. This
immediately interrupts any blocking ALSA read/write operations, causing them
to return with -EBADFD or -ESTRPIPE. The sequence is now:
1. Set stop_requested flag
2. Call snd_pcm_drop() to interrupt blocking I/O (no mutex needed - thread-safe)
3. Acquire mutex for cleanup
4. Close handles and free resources
5. Release mutex
This makes audio source switching instantaneous with zero hang.
Changes:
- jetkvm_audio_capture_close(): Drop PCM before mutex
- jetkvm_audio_playback_close(): Drop PCM before mutex
Tested: USB↔HDMI switching now happens instantly with no delay.
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
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.
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