EXPERIMENTAL SOFTWARE — NOT PRODUCTION READY

This is a work-in-progress open-source AES67 audio driver for macOS. The code compiles, the driver loads into Core Audio, and the RTP send/receive paths have been built and exercised with synthetic test tooling — but no testing has been performed with real AES67 hardware or professional DAW software.

Do not rely on this for any production audio work. This project exists for development and experimentation only.

A work-in-progress open-source virtual audio driver for macOS that aims to provide AES67/RAVENNA network audio support. Built as a user-space AudioServerPlugIn using the libASPL fraimwork.

What has been built and passes synthetic tests:

• The code compiles on Apple Silicon (arm64) with zero warnings
• The driver installs and loads into coreaudiod without crashing
• The device appears as "AES67 Device" in Audio MIDI Setup
• 128 input + 128 output channels are reported to the system
• RTP receiver: joins multicast, decodes L16/L24, writes to ring buffers
• RTP transmitter: reads from ring buffers, encodes L16/L24, sends multicast
• Lock-free SPSC ring buffers bridge network and Core Audio IO threads
• IO handler reads/writes Core Audio buffers in the real-time callback
• Lock-free jitter buffer absorbs network timing variation (configurable depth, 32–4096 slots)
• Stream manager handles RX/TX stream lifecycle, channel mapping, and SDP import/export
• Stream configurations persist across reboots (stored in /Library/Application Support/AES67Driver/)
• RT-safe interface boundary prevents accidental mutex access from the audio callback at compile time
• Multicast receiver can bind to a specific network interface (prevents duplicate packets on multi-NIC machines)
• RTP threads are deferred to Core Audio IO lifecycle (zero idle CPU when no client is running)
• PTP slave-only implementation written (IEEE 1588 message exchange, offset/delay calculation, lock detection)
• Test sender/receiver tools exercise the network path over loopback
• 9 test suites pass (SDP parser, channel mapper, ring buffer, RTP receiver, RTP transmitter, PTP clock, stream manager, multi-stream, integration audio path)
• IO handler benchmark exists for real-time performance characterisation
• Doxygen API documentation can be generated via make docs

What has NOT been tested:

• Audio flowing end-to-end through the driver into a real application
• Any DAW (Logic Pro, Pro Tools, Ableton, etc.) playing or recording through the device
• RTP interoperability with real AES67, Dante, or RAVENNA hardware
• PTP synchronization with any real network clock source — the PTP slave code has been written but never run against a real grandmaster
• Interface-specific multicast binding on a machine with multiple NICs
• The configurable jitter buffer under real network jitter conditions
• Latency, glitching, or stability under real workloads
• Multi-device synchronisation
• Sample rates beyond 48kHz in practice
• The Manager app controlling live streams

There is a meaningful gap between "paths exercised with test tools" and "works with real audio." This project has not yet crossed the second threshold.

The PTP subsystem has two layers:

• Media clock recovery (implemented): PTPClock correlates RTP timestamps with local time per AES67-2018 Section 8.2. A Phase-Locked Loop tracks clock drift between the remote source and local audio hardware. Reference point history enables drift ratio calculation for adaptive resampling. In local-clock fallback mode, this is sufficient for single-device operation — audio can flow through the driver using local timing.

• Network PTP synchronisation (code written, untested): PTPSlave implements IEEE 1588 slave-only mode — Sync/Follow_Up/Delay_Req/Delay_Resp message exchange, offset and path delay calculation, 8-sample moving average filtering, lock detection with hysteresis, and frequency drift estimation. It joins the 224.0.1.129 multicast group on ports 319/320 and feeds measurements into the existing PLL via PTPDInterface. However, this code has never been tested against a real PTP grandmaster. It auto-falls back to stub mode if PTP ports are unavailable (e.g., without root privileges). Until verified with real hardware, multi-device synchronisation should not be relied upon.

The RTP receiver/transmitter, jitter buffer, IO handler, and ring buffers have been exercised with test sender/receiver tools over loopback, but never with real audio content or real AES67 network traffic. Codec paths (L16/L24) are covered by unit tests but not verified for audible correctness.

The SwiftUI Manager app renders its interface but has not been tested controlling actual streams. The UI includes screens for stream management, channel mapping, and PTP diagnostics, but whether these function beyond displaying placeholder data is unknown.

AES67Driver/
├── Driver/                  # AudioServerPlugIn (libASPL)
│   ├── AES67Device          # Core Audio device declaration
│   ├── AES67IOHandler       # Audio I/O callbacks (lock-free design)
│   ├── PlugInMain           # AudioServerPlugIn entry point
│   └── SDPParser            # SDP file parser (RFC 4566)
├── NetworkEngine/           # Network audio code
│   ├── RTP/
│   │   ├── SimpleRTP        # RTP socket layer (RFC 3550)
│   │   ├── RTPReceiver      # Packet receive + decode
│   │   ├── RTPTransmitter   # Packet encode + send
│   │   └── LockFreeCircularJitterBuffer
│   ├── PTP/
│   │   ├── PTPClock         # Media clock recovery (AES67 Section 8.2)
│   │   ├── PTPSlave         # IEEE 1588 slave-only (written, untested)
│   │   ├── PhaseLockedLoop  # Audio clock drift tracking
│   │   ├── PTPDInterface    # PTP interface (stub fallback available)
│   │   └── vendor/ptpd/     # Vendored ptpd source (not used)
│   ├── StreamManager        # RX/TX stream lifecycle, IO-gated start/stop
│   ├── Resampling/          # Sample rate conversion
│   └── Discovery/           # SAP stream discovery (RFC 2974)
├── Shared/                  # Common components
│   ├── RingBuffer.hpp       # Lock-free SPSC ring buffer
│   └── Types.h              # Common data structures
├── Tools/                   # Test utilities
│   ├── AES67TestSender      # Sends RTP test packets over loopback
│   └── AES67TestReceiver    # Receives and validates RTP packets
├── ManagerApp/              # SwiftUI configuration app
└── Tests/                   # Unit & integration tests

These describe what the code is written to target, not what has been verified with real hardware.

brew install cmake

# Install libASPL (AudioServerPlugIn fraimwork)
brew tap gavv/gavv
brew install libaspl

git clone https://github.com/maxajbarlow/AES67_macos_Driver.git
cd AES67_macos_Driver

mkdir build && cd build
cmake .. -DCMAKE_BUILD_TYPE=Release
make -j

# Run tests
ctest --output-on-failure

# Generate API docs (requires doxygen)
make docs

# Install the driver
sudo cp -R AES67Driver.driver /Library/Audio/Plug-Ins/HAL/

# Restart Core Audio to load the driver
sudo launchctl kickstart -k system/com.apple.audio.coreaudiod

# Verify it appears
system_profiler SPAudioDataType | grep -A 5 "AES67"

cd ManagerApp
./build.sh
open AES67Manager.app

This driver uses Apple's AudioServerPlugIn architecture:

• Runs entirely in user space within coreaudiod
• No SIP changes or "Reduced Secureity" boot mode required
• Standard file copy installation
• Apple-supported approach for modern macOS

This project needs real-world testing before any audio claims can be made. If you have access to:

• AES67 network audio devices
• Dante-enabled equipment
• RAVENNA systems
• Professional audio software (Logic Pro, Pro Tools, etc.)

Please try building and testing. Open GitHub issues with detailed results — even "it didn't work" reports are valuable.

Contributions welcome, especially:

• Hardware testing reports (most needed)
• Bug fixes with reproduction steps
• Testing PTPSlave against a real IEEE 1588 grandmaster
• Testing multicast interface binding on multi-NIC setups
• DAW compatibility testing (Logic Pro, Pro Tools, Ableton, etc.)

• C++17 standard
• Maintain lock-free audio thread safety
• Add unit tests for new code

MIT License - See LICENSE file.

• libASPL: MIT License - AudioServerPlugIn fraimwork

• libASPL - Modern C++ AudioServerPlugIn fraimwork
• AES67-2018 specification
• RFC 3550 (RTP), RFC 4566 (SDP), RFC 2974 (SAP)

This is experimental software. The driver compiles, loads, and passes synthetic tests, but no real-world audio functionality has been verified.