Replace ALSA plugin layer resampling with libspeexdsp for improved audio
quality and reliability. This implementation uses direct hardware access
(hw:) instead of ALSA plugins (plughw:) and handles sample rate conversion
with SpeexDSP's high-quality sinc-based resampler.
Key changes:
- Add libspeexdsp 1.2.1 with ARM NEON optimizations to build dependencies
- Switch from plughw: to hw: device access for lower latency
- Implement conditional resampling (only when hardware rate ≠ 48kHz)
- Use SPEEX_RESAMPLER_QUALITY_DESKTOP for high-quality interpolation
- Add automatic audio dependency building in dev_deploy.sh
Quality improvements:
- Fix race condition in resampler cleanup with mutex protection
- Fix memory leak on resampler re-initialization
- Add buffer overflow validation (3840 frame limit for 192kHz)
- Improve error logging for resampling, encoding, and ALSA configuration
- Simplify code structure while maintaining all functionality
Technical details:
- Hardware negotiates actual sample rate (e.g., HDMI may vary)
- SpeexDSP converts hardware rate → 48kHz for Opus encoding
- USB Audio Gadget hardcoded to 48kHz (no resampling overhead)
- Static buffer allocation for zero allocation in hot path
- WebRTC requires 48kHz RTP clock rate per RFC 7587
USB Audio Gadget (hw:1,0) hardware only supports 48kHz for both capture
and playback due to configfs p_srate/c_srate being hardcoded. This commit
ensures both audio paths respect this hardware limitation:
- Output path: Force 48kHz when using hw:1,0, allow configurable rates for HDMI
- Input path: Always use 48kHz regardless of UI configuration
- Calculate frame size dynamically based on actual sample rate used
Also removes redundant comments that don't add debugging or maintainability value.
- Add sample rate dropdown in UI with Opus-supported rates (8k/12k/16k/24k/48kHz)
- Add sampleRate parameter to setAudioConfig RPC handler
- Validate sample rate is one of the 5 Opus-compatible values
- Configuration takes effect on next audio restart (Apply button)
Integrated latest dev branch changes including:
- Native process refactoring with gRPC architecture
- OTA update system refactor with new component-based updates
- Updated build system and dependencies
- UI improvements and bug fixes
Post-merge fixes applied:
- Remove duplicate OTA RPC function declarations (now in ota.go)
- Fix GetDefaultEDID reference to use native.DefaultEDID constant
- Fix IsUpdatePending to use otaState.IsUpdatePending() method
- Add missing OTA RPC handler registrations for new update system
All audio functionality from feat/audio-support preserved.
All dev branch functionality preserved.
The HDMI audio device can take 30-60 seconds to initialize due to
TC358743 hardware characteristics. Updated success notification in
all languages to inform users that audio will start shortly.
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.
- Set DefaultEDID in config defaults instead of empty string
- Pass config EDID to Native.Start() to fix initialization race condition
- Update DefaultEDID to MacBook-compatible value (2ch, 48kHz, 16/20/24-bit)
- Add getDefaultEDID RPC endpoint for UI to fetch backend constant
- Update UI to dynamically fetch default EDID instead of hardcoding
- Remove all EDID fallback logic now that config always has a value
- Simplify rpcGetEDID to return config value directly
This ensures the configured EDID is used from startup and eliminates
sync issues between backend constant, config, and UI.
- 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.
The sample rate cannot be configured by users - it's determined by the audio
source (HDMI device or USB gadget client). The previous UI gave the false
impression that users could select a sample rate, but the value was always
overridden by hardware detection.
Changes:
- Convert sample rate UI from dropdown to read-only display
- Show "(auto-detected from source)" label next to the value
- Remove sampleRate parameter from setAudioConfig RPC
- Update translations to clarify auto-detection
- Backend sample rate validation remains for backwards compatibility
The C code now automatically detects and adapts to whatever rate the hardware
supports, creating the Opus encoder/decoder with matching parameters to
eliminate pitch/speed distortion.
- 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 Devices.Equals() method to compare USB device configurations
- Add GetGadgetDevices() to retrieve current device state
- Skip gadget reconfiguration when device state is unchanged
- Remove 6 unused audio translation keys from all language files
- 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
- Use mono audio instead of stereo for microphone input
- Add cleanup to stop audio track on component unmount
- Explicitly set AudioTrack to nil when creation fails
- Prevent concurrent getUserMedia requests to fix toggle flakiness
- Add automatic track recovery when microphone track ends unexpectedly
- Fix auto-enable to only trigger on session start, not when toggling setting
- Ensure backend RPC is sent when auto-enable triggers
- Properly clean up old tracks before requesting new ones
- Add AudioInputAutoEnable and AudioOutputEnabled fields to backend config
- Implement RPC methods for get/set audio input auto-enable preference
- Load audio output enabled state from config on startup
- Make manual microphone toggle call backend to enable audio pipeline
- Auto-enable preference now persists across incognito sessions
- Reset microphone state to off when reaching login pages
- Fix issue where manual mic toggle required auto-enable to be on
Implement session-based microphone control with optional auto-enable:
- Microphone disabled by default, only requests permission when enabled
- Added persistent "Auto-enable Microphone" setting in audio settings
- Setting syncs to backend and works across browsers/devices
- Auto-enable respects secure context (HTTPS/localhost only)
- Refactored secure context check into reusable utility function
- Removed unused audioInputEnabled store properties
Added new audio-related UI components and improved localization for multiple languages. The SettingsItem component now supports badge links and variants, enhancing the user experience. Updated the AudioPopover to include quick audio controls for speakers and microphone, with descriptions and HTTPS-only warnings where applicable.
Temporarily remove the ability to switch between HDMI and USB audio
output sources. The application now uses USB audio (hw:1,0) exclusively
until HDMI audio capture issues are resolved.
Changes:
- Remove AudioOutputSource config field
- Remove audio source switching logic and UI
- Hardcode USB audio output device
- Remove related RPC methods
Alex P
Aveline
Adam Shiervani
Nitish Agarwal
Marc Brooks
Alex
dependabot[bot]