AES67 Linux Daemon
AES67 Linux Daemon is a Linux implementation of AES67 interoperability standard used to distribute and synchronize real time audio over Ethernet.
See https://en.wikipedia.org/wiki/AES67 for additional info.
Status
Introduction
The daemon is a Linux process that uses the Merging Technologies ALSA RAVENNA/AES67 Driver to handle PTP synchronization and RTP streams and exposes a REST interface for configuration and status monitoring.
The ALSA AES67 Driver implements a virtual ALSA audio device that can be configured using Sources and Sinks and it’s clocked using the PTP clock.
A Source reads audio samples from the ALSA playback device and sends RTP packets to a configured multicast or unicast address.
A Sink receives RTP packets from a specific multicast or unicast address and writes them to the ALSA capture device.
A user can use the ALSA capture device to receive synchronized incoming audio samples from an RTP stream and the ALSA playback device to send synchronized audio samples to an RTP stream.
The binding between a Source and the ALSA playback device is determined by the channels used during the playback and the configured Source channels map. The binding between a Sink and the ALSA capture device is determined by the channels used while recoding and the configured Sink channels map.
The driver handles the PTP and RTP packets processing and acts as a PTP clock slave to synchronize with a master clock on the specified PTP domain. All the configured Sources and Sinks are synchronized using the same PTP clock.
The daemon communicates with the driver for control, configuration and status monitoring only by using netlink sockets.
The daemon implements a REST interface to configure and monitor the Sources, the Sinks and PTP slave. See README for additional info.
It also implements SAP sources discovery and advertisement compatible with AES67 standard and mDNS sources discovery and advertisement compatible with Ravenna standard.
A WebUI is provided to allow daemon and driver configuration and monitoring. The WebUI uses the daemon REST API and exposes all the supported configuration paramaters for the daemon, the PTP slave clock, the Sources and the Sinks. The WebUI can also be used to monitor the PTP slave status and the Sinks status and to browse the remote SAP and mDNS sources.
License
AES67 daemon and the WebUI are licensed under GNU GPL.
The daemon uses the following open source:
Prerequisite
The daemon and the test have been verified starting from Ubuntu 18.04 distro onwards for ARMv7 and x86 platforms using:
- Linux kernel version >= 4.10.x
- GCC version >= 7.x / clang >= 6.x (C++17 support required)
- cmake version >= 3.7
- boost libraries version >= 1.65
- Avahi service discovery (if enabled) >= 0.7
- Freeware Advanced Audio Coder (if streamer enabled) libfaac >= 1.30
The following platforms have been used for testing:
A NanoPi NEO2 with Allwinner H5 Quad-core 64-bit Cortex A53 processor.
See Armbian NanoPi NEO2 for additional information about how to setup Ubuntu on this board.
A Mini PC N40 with Intel® Celeron® Processor N4020 , 2 Cores/2 Threads (4M Cache, up to 2.80 GHz).
See Minisforum N40 Mini PC and how to Install Ubuntu on a fanless Mini PC.
The debian-packages.sh script can be used to install all the packages required to compile and run the AES67 daemon, and the platform compatibility test.
Important CPU scaling events could affect daemon streams causing unexpected distortions, see CPU scaling events and scripts notes.
Important Starting from Linux kernel 5.10.x onwards a change in a kernel parameter is required to fix a problem with round robin scheduler causing the latency test to fail, see Real Time Scheduler Throttling.
Important PulseAudio must be disabled or uninstalled for the daemon to work properly, see PulseAudio and scripts notes.
How to build and setup the daemon
Install on your target machine (both ARM or X86) an Ubuntu Linux distribution. NO Virtual machines.
The daemon should work on all Ubuntu starting from 18.04 onward, it’s possible to use other distros.
-
Install all the required packages by running the script ubuntu-packages.sh
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execute the build.sh script to download and build the driver, the daemon and the WebUI, see script notes.
The script performs the following operations:
- checkout, patch and build the Merging Technologies ALSA RAVENNA/AES67 module
- checkout the cpp-httplib
- build and deploy the WebUI
- build the AES67 daemon
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edit the daemon configuration file daemon/daemon.conf and change the interface_name parameter to the ethernet network interface you want to use. Ex: eth0
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install the driver by executing from the main folder:
(cd 3rdparty/ravenna-alsa-lkm/driver && sudo insmod ./MergingRavennaALSA.ko)
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start the daemon by executing from the main folder:
cd daemon
./aes67-daemon -c daemon.conf
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use the daemon option -a if you want to bind the daemon HTTP server to another address. For example -a 0.0.0.0 to bind the daemon HTTP server to all the local network interfaces.
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open a browser and load the daemon configuration WebUI at http://[your_ip]:8080
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on the WebUI use the Browser tab to see the other ASE67 Sources advertised on the network.
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on the WebUI use the Sinks tab to create a receiver by specifying a remote Source manually or by selecting one of the remote Source discovered (check the Use SDP option to enable this).
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on the WebUI use the Sources tab to create a daemon source
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on the WebUI use the PTP tab to check the status of the daemon PTP slave. Receiving or sending audio via the ALSA RAVENNA device is only possible if the PTP slave status is locked.
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playback on the RAVENNA ALSA device to send audio via a Source, for example for a stereo source use:
speaker-test -D plughw:RAVENNA -r 48000 -c 2 -t sine
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record from the RAVENNA ALSA device to receive audio via a Sink, for example to record to a file a stereo sink use:
arecord -D plughw:RAVENNA -c 2 -f cd -d 30 -r 48000 -t wav sink.wav
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to run interoperability tests with other AES67 devices, for example with Dante devices use instructions below.
Devices and interoperability tests
See Devices and interoperability tests with the AES67 daemon
Support for ST-2022-7
Starting from the daemon version 3.0 and driver version 2.0 support for ST-2022-7 was added.
This feature is automatically enabled when 2 interfaces are configured via the daemon interface_name parameter.
When ST-2022-7 is enabled:
- the driver’s PTP slave receives packets from both interfaces, and a master clock is automatically elected.
- for a Source, outgoing packets are transmitted through both interfaces.
- for a Sink, packets are received from both interfaces, and the incoming streams are merged accordingly.
- AES67 control (SAP and mDNS) runs on all interfaces.
Beginning with daemon version 3.1, both the daemon and the WebUI support specifying a separate address and port for the secondary interface (daemon parameters rtp_mcast_base_sec and rtp_port_sec).
When ST-2022-7 is enabled the daemon uses these values for the Sources and advertises the redundant multicast audio flow in the associated SDP file.
For example:
v=0
o=- 2831159553 317021570 IN IP4 192.168.1.17
s=Daemon a8c01101 ALSA Source 0
t=0 0
a=group:DUP 1 2
m=audio 5004 RTP/AVP 98
c=IN IP4 239.1.0.1/15
a=source-filter: incl IN IP4 239.1.0.1 192.168.1.17
a=rtpmap:98 L24/48000/2
a=sync-time:0
a=framecount:48
a=ptime:1
a=mediaclk:direct=0
a=clock-domain:PTPv2 0
a=ts-refclk:ptp=IEEE1588-2008:00-1D-C1-FF-FE-50-36-33:0
a=recvonly
a=mid:1
m=audio 5006 RTP/AVP 98
c=IN IP4 239.1.0.1/15
a=source-filter: incl IN IP4 239.1.0.1 192.168.1.18
a=rtpmap:98 L24/48000/2
a=sync-time:0
a=framecount:48
a=ptime:1
a=mediaclk:direct=0
a=clock-domain:PTPv2 0
a=ts-refclk:ptp=IEEE1588-2008:00-1D-C1-FF-FE-50-36-33:0
a=mid:2
If a specific RTP address is defined when the Source is created, that address will be used for both the primary and secondary interfaces, replacing the default addresses.
For a Sink, when ST-2022-7 is enabled, the daemon retrieves the audio media redundant multicast flow address and port from the remote Source’s SDP file. If these are not specified the daemon uses the address and port specified for the primary.
The primary audio stream is always transmitted and received via the primary interface, while the redundant audio stream is always transmitted and received via the secondary interface.
A single interface can be configured both as primary and secondary at the same time.
HTTP Streamer
The HTTP Streamer was introduced with the daemon version 2.0 and it is used to receive AES67 audio streams via HTTP file streaming.
Starting from the daemon version 2.2 the Streamer supports live HTTP streaming compatible with VideoLAN.
The HTTP Streamer can be enabled via the streamer_enabled daemon parameter.
When the Streamer is active the daemon starts capturing the configured Sinks up to the maximum number of channels configured by the streamer_channels parameters.
The captured PCM samples are split into streamer_files_num files of streamer_file_duration duration (in seconds) for each sink, compressed using AAC LC codec and served via HTTP.
The HTTP streamer requires the libfaac-dev package to compile.
When the HTTP Streamer is enabled and one or more Sinks are configured, it’s possible to receive a HTTP Live stream encoded using AAC LC at the following URL: http://[daemon_ip:port]/api/streamer/stream/:sinkId
Please note that since the HTTP Streamer uses the RAVENNA ALSA device for capturing it’s not possible to use such device for other audio captures.
AES67 USB Receiver and Transmitter
See Use your board as AES67 USB Receiver and Transmitter
Repository content
This directory contains the AES67 daemon source code.
The daemon can be cross-compiled for multiple platforms and implements the following functionalities:
- communication and configuration of the ALSA RAVENNA/AES67 device driver
- control and configuration of up to 64 multicast and unicast sources and sinks using the ALSA RAVENNA/AES67 driver via netlink
- session handling and SDP parsing and creation
- HTTP REST API for the daemon control and configuration
- SAP sources discovery, update and advertisement compatible with AES67 standard
- mDNS sources discovery and advertisement (using Linux Avahi) compatible with Ravenna standard
- RTSP client and server to retrieve, return and update SDP files via DESCRIBE and ANNOUNCE methods according to Ravenna standard
- IGMP handling for SAP, PTP and RTP sessions
- Integration with systemd watchdog monitoring (from daemon release v1.6)
See the README file in this directory for additional information about the AES67 daemon configuration and the daemon HTTP REST API.
The directory also contains the daemon tests in the tests subdirectory.
Daemon tests can be executed via a Docker container by using a fake version of the daemon driver manager.
This was implemented to perfom automated execution of the tests via a GitHub workflow.
To build the Docker image for the daemon regression tests run:
docker build --progress=plain -f ./Dockerfile.daemon_tests -t aes67-daemon-tests .
To run the tests:
docker run aes67-daemon-tests
This directory contains the AES67 daemon WebUI configuration implemented using React.
With the WebUI a user can do the following operations:
- change the daemon configuration, this causes a daemon restart
- edit PTP clock slave configuration and monitor PTP slave status
- add and edit RTP Sources
- add, edit and monitor RTP Sinks
- browser remote SAP and mDNS RTP sources
This directory is used to download the 3rdparty open source.
The aes67-daemon branch of ravenna-alsa-lkm repository contains the ALSA RAVENNA/AES67 module code plus all the patches applied to it.
The ALSA RAVENNA/AES67 kernel module is responsible for:
- registering as an ALSA driver
- generating and receiving RTP audio packets
- PTP slave operations and PTP driven interrupt loop
- netlink communication between user and kernel
The following patches have been applied to the original module:
- set of patches to improve driver stability and performance, see 367c166. Added audio_cpu_affinity module parameter to have the audio timer pinned to a specific CPU (from driver 2.1).
- added support for ST-2022-7 (from driver version 2.0). This version breaks compatibility with the older and requires a new daemon. See issue 248
- patch to update the grand master clock ID of the current master clock when it gets updated in the ANNOUNCE messages (from driver version v1.18). See driver issue 34
- patch to fix the PTP master sync timeout and to have a less restrictive spin lock (from driver version v1.17). See issue 246
- patch to fix a kernel warning with Linux Kernel v6.18 and above (from driver version v1.16). See driver issue 32
- patch to compile with Linux Kernel v6.15 and above (from driver version v1.15)
- patch to properly release the network device, that on some Linux Kernels was preventing device reboot and shutdown (from driver v1.14)
- patch to use ALSA managed buffer allocation and compile with Linux Kernel v6.12 and above (from driver version v1.13)
- patch to compile with Linux Kernel v6.11 and above (from driver version v1.11)
- patch to support sample rates 192Khz and 384Khz (from driver version v1.7)
- patch to compile with Linux Kernel v6.2 and above (from driver version v1.7)
- patch to make the PTP slave status change from locked to unlocked if no announcement messages are received from the Master for more than 5 seconds (from driver version v1.6). See issue 87.
- patch to fix issue causing the usage of wrong Sources and Sinks buffer offsets in case of reconfiguration or daemon restart (from driver version v1.5). See issue 55.
- patch to enable the configuration of smaller ALSA buffer sizes to reduce end-to-end latency up to 8ms (from driver version v1.4). See platform latency test and See issue 53
- patch to remove unsupported non-interleved access from driver capabilities. This enables compatibility with JACK audio (from driver version v1.3)
- patch to enable the usage of a PTP master clock on the daemon host (from driver version v1.1)
- patch to rework the driver PCM interface and to simplify and unify handling of read-write interlaved and memory mapped access modes
- patch to remove user space transfer handling from convert functions
- patch to add support for AM824 (L32) codec
- patch to initialize playback and capture buffers
- patch to enable mono channel support to playback and capture ALSA devices
- patch to enable direct PCM transfer mode instead of indirect to enable the use of ALSA plugins
- patch to enable independent playback and capture interrupt startup
- patch to disable UDP checksum check for PTP packets
- patch to enable support for ARM 32bit and 64bit platforms
- patch to enable the network loopback device
- patch to compile with Linux Kernel v5
See ALSA RAVENNA/AES67 Driver README for additional information about the Merging Technologies module and for proper Linux Kernel configuration and tuning.
This directory contains systemd configuration files for the daemon.
The daemon integrates with systemd watchdog.
To enable it recompile the daemon with the CMake option -DWITH_SYSTEMD=ON
You can install the daemon under systemd by using the script systemd/install.sh:
cd systemd
sudo ./install.sh
Before starting the daemon edit /etc/daemon.conf and make sure the interface_name parameter is set to your ethernet interface.
To start the daemon use:
sudo systemctl start aes67-daemon
To stop it use:
sudo systemctl stop aes67-daemon
The daemon requires the MergingRavennaALSA module to run.
You can usally install the module using the following commands:
cd 3rdparty/ravenna-alsa-lkm/driver
sudo make modules_install
If this doesn’t work because you miss kernel certificate follow the instructions at:
No OpenSSL sign-file signing_key.pem
On some distros you may also need to create a module load configuration file with:
echo 'MergingRavennaALSA' | sudo tee /etc/modules-load.d/MergingRavennaALSA.conf
Finally use the command to load the modules:
sudo depmod -a
and reboot with:
sudo reboot
test directory
This directory contains the files used to run the daemon platform compatibility test on the network loopback interface. The test is described below.
Before attempting to use the AES67 daemon on a specific host or board it’s highly recommended to run the platform test.
The platform test runs a playback and recording of an RTP session on the loopback network device using the ALSA RAVENNA/AES67 modules and checks that no audio samples get corrupted or lost.
This test can be executed using the run_test.sh script. See script notes.
The script allows a user to test a specific configuration and it can be used to ensure that the daemon will be able to operate smoothly with such config on the current platform.
Usage run_test.sh sample_format sample_rate channels duration
sample_format can be one of S16_LE, S24_3LE, S32_LE
sample_rate can be one of 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000
channels can be one of 2, 4, 6, up to 64
duration is in the range 1 to 10 minutes
For example to test the typical AES67 configuration run:
./run_test.sh S24_3LE 48000 2 5
The test performs the following operations:
- check that all the required executables are available
- stop the running daemon instances and remove the ALSA RAVENNA/AES67 module
- compile the test tools under the test folder
- validates the input parameters, prepare the raw input file to be played ./test/test.raw and the configuration files under the test folder
- stop PulseAudio (if installed). This opens and keeps busy the ALSA playback and capture devices causing problems. See PulseAudio.
- install the ALSA RAVENNA/AES67 module
- start the ptp4l as master clock on the network loopback interface
- start the AES67 daemon and create a source and a sink according to the status file created in the test directory
- wait for the Ravenna driver PTP slave to synchronize
- start recording on the configured ALSA sink for the specified period tof time to the raw file ./test/sink_test.raw
- start playing the test file created ./test/test.raw on the configured ALSA source
- wait for the recording and the playback to complete
- check that the recorded file contains the expected audio samples sequence
- terminate ptp4l and the AES67 daemon
- print the test result that can be either OK or failed at (location)
If the test result is OK it means that the selected configuration can run smoothly on your platform.
A 64 channels configuration was succesfully tested on the Mini PC with Intel Celeron N4020 processor.
A 24 channels configuration was succesfully tested on the NanoPi NEO2 board.
If the test reports a failure you may try to stop all the possible additional loads running on the host and repeat it.
If after this the test fails systematically it means you cannot achieve a good reliability with the specified configuration.
In this case you may try to configure a different driver timer basic tick period in the daemon configuration file (parameter tic_frame_size_at_1fs in test/daemon.conf).
By default this parameter is set to 48 (1ms latency) and the valid range is from 48 to 480 with steps of 48.
Note that higher values of this parameter (values above 48) lead to higher packets processing latency and this breaks the compatibility with certain devices.
The platform latency test can be used to measure the minimum end-to-end latency that can be achieved on a specific platform.
The test is based on the same setup used for the platform compatibility test where a Sink is configured to receive audio samples from a Source
both running on the same device using the network loopback interface.
A test application plays audio samples on the RAVENNA playback device and measures the time till the samples are received on the RAVENNA capture device.
The test setup and the end-to-end latency measured are shown in the picture below:
The test can be executed using the run_latency_test.sh script. See script notes.
The script allows a user to test the latency on a specific configuration and it can be used to ensure that a specific ALSA buffer size can be used:
Usage run_latency_test.sh sample_format sample_rate channels duration frames
sample_format can be one of S16_LE, S24_3LE, S32_LE
sample_rate can be one of 44100, 48000, 88200, 96000, 192000, 384000
channels can be one of 2, 4, 6, up to 64
duration of the test in seconds
frames buffer size in frames
The specified buffer size in frames starts from tic_frame_size_at_1fs * 2 (96 by default) in steps of tic_frame_size_at_1fs.
For example, to test the typical AES67 configuration for 1 minute and a buffer size of 96 frames run:
./run_latency_test.sh S24_3LE 48000 2 60 96
If no underrun errors occurred during the test the requested buffer size can be used and the end-to-end latency measured is printed at the end:
Trying latency 96 frames, 2000.000us, 2.000000ms (500.0000Hz)
Success
Playback:
*** frames = 480096 ***
state : RUNNING
trigger_time: 170.196886
tstamp : 0.000000
delay : 96
avail : 0
avail_max : 48
Capture:
*** frames = 480000 ***
state : RUNNING
trigger_time: 170.196887
tstamp : 0.000000
delay : 0
avail : 0
avail_max : 48
Maximum read: 48 frames
Maximum read latency: 1000.000us, 1.000000ms (1000.0000Hz)
Playback time = 170.196886, Record time = 170.196887, diff = -1
End to end latency: 5.999 msecs
Terminating processes ...
daemon exiting with code: 0
The previous test was run on a NanoPi NEO2 board with Ubuntu distro.
A 64 channels configuration was succesfully tested on the Mini PC with Intel Celeron N4020 processor.
A 24 channels configuration was succesfully tested on the NanoPi NEO2 board.
In case underrun happened the status reported is:
state : XRUN
and the specified buffer size cannot be used.
Important Starting from Linux kernel 5.10.x onwards a change in a kernel parameter is required to fix a problem with round robin scheduler causing the latency test to fail, see Real Time Scheduler Throttling.
Run the daemon tests
To run daemon tests install the ALSA RAVENNA/AES67 kernel module with:
sudo insmod 3rdparty/ravenna-alsa-lkm/driver/MergingRavennaALSA.ko
setup the kernel parameters with:
sudo sysctl -w net/ipv4/igmp_max_memberships=66
make sure that no instances of the aes67-daemon are running, enter the tests subdirectory and run:
./daemon-test -p
Notes
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All the scripts in this repository are provided as a reference to help setting up the system and run the platform compatibility test.
They have been tested starting from Ubuntu 18.04 distros onwards.
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Starting from Linux kernel 5.10.x onwards a change in a kernel parameter is required to fix a problem with round robin scheduler causing the latency test to fail, see issue 79 and alsa-lib issue-285.
Set the total bandwidth available to all real-time tasks to 100% of the CPU with:
sysctl -w kernel.sched_rt_runtime_us=1000000
By default this value is set to 95%.
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CPU scaling events could have an impact on daemon streams causing unexpected distortion for a few seconds, see issue 96.
Before running the daemon make sure you disable CPU scaling events:
Check if CPU scaling is enabled with:
cat /proc/sys/kernel/perf_cpu_time_max_percent
If result is not 0, (it’s usually set to 25) set it to 0 with:
sudo sysctl -w kernel.perf_cpu_time_max_percent=0
You may also want to review the current CPU scaling governor with (cpu0 in this case):
cat /sys/devices/system/cpu/cpu0/cpufreq/scaling_governor
See https://www.kernel.org/doc/Documentation/cpu-freq/governors.txt for dditional info.
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PulseAudio can create instability problems.
Before running the daemon verify that PulseAudio is not running with:
ps ax | grep pulseaudio
In case it’s running try to execute the following script to stop it:
daemon/scripts/disable_pulseaudio.sh
If after this the process is still alive consider one of these two solutions and reboot the system afterwards:
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Uninstall it completely with:
sudo apt-get remove pulseaudio
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Disable it by renaming the executable with:
sudo mv /usr/bin/pulseaudio /usr/bin/_pulseaudio
Other methods to disable PulseAudio may fail and just killing it is not enough since it gets immediately re-spawned.
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