A tool to create audio processing pipelines for applications such as active crossovers or room correction. It is written in Rust to benefit from the safety and elegant handling of threading that this language provides.
Supported platforms: Linux, macOS, Windows.
Audio data is captured from a capture device and sent to a playback device. Alsa, PulseAudio, Jack, Wasapi and CoreAudio are currently supported for both capture and playback.
The processing pipeline consists of any number of filters and mixers. Mixers are used to route audio between channels and to change the number of channels in the stream. Filters can be both IIR and FIR. IIR filters are implemented as biquads, while FIR use convolution via FFT/IFFT. A filter can be applied to any number of channels. All processing is done in chunks of a fixed number of samples. A small number of samples gives a small in-out latency while a larger number is required for long FIR filters. The full configuration is given in a yaml file.
CamillaDSP is distributed under the GNU GENERAL PUBLIC LICENSE Version 3.
This includes the following disclaimer:
- Disclaimer of Warranty.
THERE IS NO WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
- Limitation of Liability.
IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
In short this means that the user is responsible for any damage resulting from using this program. It does not matter if the damage is caused by incorrect usage or a bug in the software.
- Background
- How it works
- System requirements
- Usage example: crossover for 2-way speakers
- Dependencies
- Companion libraries and tools
- GUI
- Build with standard features
- Customized build
- Optimize for your system
- Building on Windows and macOS
- The YAML format
- Title and description
- Volume control
- Devices
- Resampling
- Mixers
- Filters
- Processors
- Pipeline
- Export filters from REW
- Visualizing the config
The purpose of CamillaDSP is to enable audio processing with combinations of FIR and IIR filters. This functionality is available in EqualizerAPO, but for Windows only. For Linux the best known FIR filter engine is probably BruteFIR, which works very well but doesn't support IIR filters. The goal of CamillaDSP is to provide both FIR and IIR filtering for Linux, Windows and macOS, to be stable, fast and flexible, and be easy to use and configure.
- BruteFIR: https://torger.se/anders/brutefir.html
- EqualizerAPO: https://sourceforge.net/projects/equalizerapo/
- The IIR filtering is heavily inspired by biquad-rs: https://github.com/korken89/biquad-rs
The audio pipeline in CamillaDSP runs in three separate threads. One thread handles capturing audio, one handles the playback, and one does the processing in between. The capture thread passes audio to the processing thread via a message queue. Each message consists of a chunk of audio with a configurable size. The processing queue waits for audio messages, processes them in the order they arrive, and passes the processed audio via another message queue to the playback thread. There is also a supervisor thread for control. This chart shows the most important parts:
The capture thread reads a chunk samples from the audio device in the selected format. It then converts the samples to 64-bit floats (or optionally 32-bit). If resampling is enabled, the audio data is sent to the resampler. At the end, the chunk of samples is packed as a message that is then posted to the input queue of the processing thread. After this the capture thread returns to reading the next chunk of samples from the device.
The processing thread waits for audio chunk messages to arrive in the input queue. Once a message arrives, it's passed through all the defined filters and mixers of the pipeline. Once all processing is done, the audio data is posted to the input queue of the playback device.
The playback thread simply waits for audio messages to appear in the queue. Once a message arrives, the audio data is converted to the right sample format for the device, and written to the playback device. The Alsa playback device supports monitoring the buffer level of the playback device. This is used to send requests for adjusting the capture speed to the supervisor thread, on a separate message channel.
The supervisor monitors all threads by listening to their status messages. The requests for capture rate adjust are passed on to the capture thread. It's also responsible for updating the configuration when requested to do so via the websocket server or a SIGHUP signal.
The websocket server launches a separate thread to handle each connected client. All commands to change the config are sent to the supervisor thread.
CamillaDSP runs on Linux, macOS and Windows. The exact system requirements are determined by the amount of processing the application requires, but even relatively weak CPUs like Intel Atom have much more processing power than most will need.
In general, a 64-bit CPU and OS will perform better.
A few examples, done with CamillaDSP v0.5.0:
-
A Raspberry Pi 4 doing FIR filtering of 8 channels, with 262k taps per channel, at 192 kHz. CPU usage about 55%.
-
An AMD Ryzen 7 2700u (laptop) doing FIR filtering of 96 channels, with 262k taps per channel, at 192 kHz. CPU usage just under 100%.
Both 64 and 32 bit architectures are supported. All platforms supported by the Rustc compiler should work.
Pre-built binaries are provided for:
- x86_64 (almost all PCs)
- armv7 (32-bit arm, for example a Raspberry Pi 2,3,4 with a 32-bit OS)
- aarch64 (64-bit arm, for example Raspberry Pis running a 64 bit OS)
An x86_64 CPU and the 64-bit version of Windows is recommended. Any x86_64 CPU will likely be sufficient.
Pre-built binaries are provided for 64-bit systems.
CamillaDSP can run on both Intel and Apple Silicon macs. Any reasonably recent version of MacOS should work.
Pre-built binaries are provided for both Intel and Apple Silicon
A crossover must filter all sound being played on the system. This is possible with both PulseAudio and Alsa by setting up a loopback device (Alsa) or null sink (Pulse) and setting this device as the default output device. CamillaDSP is then configured to capture from the output of this device and play the processed audio on the real sound card.
See the tutorial for a step-by-step guide.
These are the key dependencies for CamillaDSP.
- https://crates.io/crates/alsa - Alsa audio backend
- https://crates.io/crates/clap - Command line argument parsing
- https://crates.io/crates/cpal - Jack audio backend
- https://crates.io/crates/libpulse-simple-binding - PulseAudio audio backend
- https://crates.io/crates/realfft - Wrapper for RustFFT that speeds up FFTs of real-valued data
- https://crates.io/crates/rustfft - FFT used for FIR filters
- https://crates.io/crates/rubato - Sample rate conversion
- https://crates.io/crates/serde_yaml - Config file reading
- https://crates.io/crates/tungstenite - Websocket server
These projects are part of the CamillaDSP family:
- https://github.com/HEnquist/pycamilladsp - Library for communicating with CamillaDSP over websocket.
- https://github.com/HEnquist/pycamilladsp-plot - Plotting and visualization of configurations.
- https://github.com/HEnquist/camilladsp-config - Example configurations for things like running CamillaDSP as a systemd service.
CamillaGUI is a user interface for CamillaDSP that is accessed via a web browser.
The easiest way to install CamillaDSP is to download a pre-built binary. Binaries for each release are available for the most common systems. See the "Releases" page. To see the files click "Assets".
These are compressed files containing a single executable file that is ready to run.
The following configurations are provided:
| Filename | Description | Backends |
|---|---|---|
camilladsp-linux-amd64.tar.gz |
Linux on 64-bit Intel or AMD CPU | Alsa, Pulseaudio |
camilladsp-linux-armv7.tar.gz |
Linux on Armv7 with Neon (32-bit), intended for Raspberry Pi 2 and up but should also work on others | Alsa |
camilladsp-linux-aarch64.tar.gz |
Linux on Armv8 (64-bit), intended for Raspberry Pi 3 and up, but should also work on others | Alsa |
camilladsp-macos-amd64.tar.gz |
macOS on 64-bit Intel CPU | CoreAudio |
camilladsp-macos-aarch64.tar.gz |
macOS on Apple silicon | CoreAudio |
camilladsp-windows-amd64.zip |
Windows on 64-bit Intel or AMD CPU | Wasapi |
All builds include the Websocket server.
The .tar.gz-files can be uncompressed with the tar command:
tar -xvf camilladsp-linux-amd64.tar.gzUse recent stable versions of rustc and cargo. The minimum rustc version is 1.61.0.
The recommended way to install rustc and cargo is by using the "rustup" tool. This tool works on all supported platforms (Linux, macOS and Windows). Get it here: https://rustup.rs/
For Windows you also need the "Build Tools for Visual Studio". Get them from here: https://aka.ms/buildtools
When building on Linux the Alsa backend is always enabled. Similarly, building on Windows always enables the Wasapi backend. And building on macOS always enables the CoreAudio backend.
By default both the PulseAudio and Jack backends are disabled, but they can be enabled if desired. Leaving them disabled also means that the corresponding system Jack/Pulse packages aren't needed.
By default the internal processing is done using 64-bit floats. There is a possibility to switch this to 32-bit floats. This might be useful for speeding up the processing when running on a 32-bit CPU (or a 64-bit CPU running in 32-bit mode), but the actual speed advantage has not been evaluated. Note that the reduction in precision increases the numerical noise.
CamillaDSP includes a Websocket server that can be used to pass commands to the running process. This feature is enabled by default, but can be left out. The feature name is "websocket". For usage see the section "Controlling via websocket".
The default FFT library is RustFFT, but it's also possible to use FFTW. This is enabled by the feature "FFTW". When the chunksize is a power of two, like 1024 or 4096, then FFTW and RustFFT are very similar in speed. But if the chunksize is a "strange" number like a large prime, then FFTW can be faster. FFTW is a much larger and more complicated library, so using FFTW is only recommended if you for some reason can't use an "easy" chunksize and this makes RustFFT too slow.
These instructions assume that the linux distribution used is one of Fedora, Debian, Ubunty or Arch. They should also work also work on distributions closely related to one of these, such as Manjaro (Arch), or Raspberry Pi OS (Debian).
There are many others, including some specialized distributions for example targeting audio playback. These often come with a slimmed down set of preinstalled packages. Compiling CamillaDSP on one of these may require installing several more packages than the ones listed here. If possible, it's recommended to use a pre-built binary on these systems.
- Install pkg-config (very likely already installed):
-
- Fedora:
sudo dnf install pkgconf-pkg-config
- Fedora:
-
- Debian/Ubuntu etc:
sudo apt-get install pkg-config
- Debian/Ubuntu etc:
-
- Arch:
sudo pacman -S cargo pkg-config
- Arch:
- Install Alsa dependency:
-
- Fedora:
sudo dnf install alsa-lib-devel
- Fedora:
-
- Debian/Ubuntu etc:
sudo apt-get install libasound2-dev
- Debian/Ubuntu etc:
-
- Arch:
sudo pacman -S alsa-lib
- Arch:
- Install OpenSSL dependency:
-
- Fedora:
sudo dnf install openssl openssl-devel
- Fedora:
-
- Debian/Ubuntu etc:
sudo apt-get install openssl libssl-dev
- Debian/Ubuntu etc:
-
- Arch:
sudo pacman -S openssl
- Arch:
- Clone the repository
- Build with standard options:
cargo build --release -
- see below for other options
- The binary is now available at ./target/release/camilladsp
- Optionally install with
cargo install --path . -
- Note: the
installcommand takes the same options for features as thebuildcommand.
- Note: the
All the available options, or "features" are:
pulse-backend: PulseAudio support.cpal-backend: Used for Jack support (automatically enabled when needed).jack-backend: Jack support.bluez-backend: Bluetooth support via BlueALSA (Linux only).websocket: Websocket server for control.secure-websocket: Enable secure websocket, also enables thewebsocketfeature.FFTW: Use FFTW instead of RustFFT.32bit: Perform all calculations with 32-bit floats (instead of 64).
The websocket feature is included in the default features, meaning it will be enabled if you don't specify anything.
Cargo doesn't allow disabling a single default feature,
but you can disable the whole group with the --no-default-features flag.
Then you have to manually add all the ones you want.
Example 1: You want websocket, pulse-backend and FFTW. The first one is included by default so you only need to add FFTW and pulse-backend:
cargo build --release --features FFTW --features pulse-backend
(or)
cargo install --path . --features FFTW --features pulse-backend
Example 2: You want 32bit and FFTW. Since you don't want websocket you have to disable the defaults:
cargo build --release --no-default-features --features FFTW --features 32bit
(or)
cargo install --path . --no-default-features --features FFTW --features 32bit
The pulse-backend feature requires PulseAudio and its development files. To install:
- Fedora:
sudo dnf install pulseaudio-libs-devel - Debian/Ubuntu etc:
sudo apt-get install libpulse-dev - Arch:
sudo pacman -S libpulse
The jack-backend feature requires jack and its development files. To install:
- Fedora:
sudo dnf install jack-audio-connection-kit jack-audio-connection-kit-devel - Debian/Ubuntu etc:
sudo apt-get install jack libjack-dev - Arch:
sudo pacman -S jack
By default Cargo builds for a generic system, meaning the resulting binary might not run as fast as possible on your system. This means for example that it will not use AVX on an x86-64 CPU, or NEON on a Raspberry Pi.
To make an optimized build for your system, you can specify this in your Cargo configuration file. Or, just set the RUSTFLAGS environment variable by adding RUSTFLAGS='...' in from of the "cargo build" or "cargo install" command.
Make an optimized build on x86-64:
RUSTFLAGS='-C target-cpu=native' cargo build --release
On a Raspberry Pi also state that NEON should be enabled:
RUSTFLAGS='-C target-feature=+neon -C target-cpu=native' cargo build --release
The platform-specific backends are always enabled when building on Windows and macOS. The recommended build command is:
macOS:
RUSTFLAGS='-C target-cpu=native' cargo build --release
Windows (cmd.exe command prompt):
set RUSTFLAGS=-C target-cpu=native
cargo build --release
Windows (PowerShell):
$env:RUSTFLAGS="-C target-cpu=native"
cargo build --release
On macOS both the PulseAudio and FFTW features can be used. The necessary dependencies can be installed with brew:
brew install fftw
brew install pkg-config
brew install pulseaudio
The FFTW feature can also be used on Windows. There is no need to install anything extra.
The command is simply:
camilladsp /path/to/config.yml
This starts the processing defined in the specified config file. The config is first parsed and checked for errors. This first checks that the YAML syntax is correct, and then checks that the configuration is complete and valid. When an error is found it displays an error message describing the problem. See more about the configuration file below.
Starting with the --help flag prints a short help message:
> camilladsp.exe --help
CamillaDSP 2.0.0
Henrik Enquist <[email protected]>
A flexible tool for processing audio
Built with features: websocket
Supported device types:
Capture: File, Stdin, Wasapi
Playback: File, Stdout, Wasapi
USAGE:
camilladsp.exe [FLAGS] [OPTIONS] <configfile>
FLAGS:
-m, --mute Start with the volume control muted
-c, --check Check config file and exit
-h, --help Prints help information
-V, --version Prints version information
-v Increase message verbosity
-w, --wait Wait for config from websocket
OPTIONS:
-s, --statefile <statefile> Use the given file to persist the state
-o, --logfile <logfile> Write logs to file
-l, --loglevel <loglevel> Set log level [possible values: trace, debug, info, warn, error, off]
-a, --address <address> IP address to bind websocket server to
-g, --gain <gain> Set initial gain in dB for the volume control
-p, --port <port> Port for websocket server
-n, --channels <channels> Override number of channels of capture device in config
-e, --extra_samples <extra_samples> Override number of extra samples in config
-r, --samplerate <samplerate> Override samplerate in config
-f, --format <format> Override sample format of capture device in config [possible values: S16LE,
S24LE, S24LE3, S32LE, FLOAT32LE, FLOAT64LE]
ARGS:
<configfile> The configuration file to use
Most flags have a long and a short form. For example --port 1234 and -p1234 are equivalent.
If the --check flag is given, the program will exit after checking the configuration file.
Use this if you only want to verify that the configuration is ok, and not start any processing.
The default logging setting prints messages of levels "error", "warn" and "info".
This can be changed with the loglevel option.
Setting this to for example warn will print messages of level warn and above, but suppress the lower levels of info, debug and trace.
Alternatively, the log level can be changed with the verbosity flag.
By passing the verbosity flag once, -v, debug messages are enabled.
If it's given twice, -vv, it also prints trace messages.
The log messages are normally written to the terminal via stderr, but they can instead be written to a file by giving the --logfile option.
The argument should be the path to the logfile. If this file is not writable, CamillaDSP will panic and exit.
The --statefile option is used to give a path to a file where CamillaDSP will save the config file path, and the volume and mute settings.
On startup, these values will be read from the statefile.
The values in the file will then be kept updated whenever they change.
If the configfile argument is given, then this will be used instead of the value from the statefile.
Similarly, the --gain and --mute options also override the values in the statefile.
Use this feature with caution! The volume setting given in the statefile will be applied immediately when CamillaDSP starts processing.
In systems that have a gain structure such that a too high volume setting can damage equipment or ears,
it is recommended to always use the --gain option to set the volume to start at a safe value.
To enable the websocket server, provide a port number with the --port option. Leave it out, or give 0 to disable.
By default the websocket server binds to the address 127.0.0.1 which means it's only accessible locally (to clients running on the same machine).
If it should be also available to remote machines, give the IP address of the interface where it should be available with the --address option.
Giving 0.0.0.0 will bind to all interfaces.
If CamillaDSP was built with the "secure-websocket" feature, it has two additional options --cert and --pass.
These are used to provide an identity, to enable secure websocket connections.
See the websocket readme for more details.
If the "wait" flag, --wait is given, CamillaDSP will start the websocket server and wait for a configuration to be uploaded.
Then the config file argument must be left out.
There are a few options to override values in the loaded config file.
Giving these options means the provided values will be used instead of the values in any loaded configuration.
To change the values, CamillaDSP has to be restarted.
If the config file has resampling disabled, then overriding the samplerate will change the samplerate parameter.
But if resampling is enabled, it will instead change the capture_samplerate parameter.
If then enable_rate_adjust is false and capture_samplerate=samplerate, then resampling will be disabled.
When overriding the samplerate, two other parameters are scaled as well.
Firstly, the chunksize is multiplied or divided by integer factors to try to keep the pipeline running at a constant number of chunks per second.
Secondly, the value of extra_samples is scaled to give the extra samples the same duration at the new samplerate.
But if the extra_samples override is used, the given value is used without scaling it.
The --gain option can accept negative values, but this requires a little care since the minus sign can be misinterpreted as another option.
It works as long as there is no space in front of the minus sign.
These work (for a gain of +/- 12.3 dB):
-g12.3
-g 12.3
--gain 12.3
--gain=12.3
-g-12.3
--gain=-12.3
These will NOT work:
-g -12.3
--gain -12.3
These are the exit codes CamillaDSP will give:
| Exit code | Meaning |
|---|---|
| 0 | Normal exit, no error |
| 101 | Invalid config file, see the error message for details |
| 102 | Error from DSP process, see the error message for details |
The configuration can be reloaded without restarting by sending a SIGHUP to the camilladsp process. This will reload the config and if possible apply the new settings without interrupting the processing. Note that for this to update the coefficients for a FIR filter, the filename of the coefficients file needs to change.
See the separate readme for the websocket server
The goal is to insert CamillaDSP between applications and the sound card. The details of how this is achieved depends on which operating system and which audio API is being used. It is also possible to use pipes for apps that support reading or writing audio data from/to stdout.
These backends are supported on all platforms.
Audio can be read from a file or a pipe using the File device type.
This can read raw interleaved samples in most common formats.
To instead read from stdin, use the Stdin type.
This makes it possible to pipe raw samples from some applications directly to CamillaDSP, without going via a virtual soundcard.
Jack is most commonly used with Linux, but can also be used with both Windows and MacOS.
The Jack support of the current CamillaDSP version (v0.6.0 at the time of writing) should be considered experimental.
The jack server must be running.
Set device to "default" for both capture and playback.
The sample format is fixed at 32-bit float (FLOAT32LE).
The samplerate must match the samplerate configured for the Jack server.
CamillaDSP will show up in Jack as "cpal_client_in" and "cpal_client_out".
See the separate readme for Wasapi.
See the separate readme for CoreAudio.
Linux offers several audio APIs that CamillaDSP can use.
See the separate readme for ALSA.
PulseAudio provides a null-sink that can be used to capture audio from applications. To create a null sink type:
pacmd load-module module-null-sink sink_name=MySink
This device can be set as the default output, meaning any application using PulseAudio will use it. The audio sent to this device can then be captured from the monitor output named "MySink.monitor".
All available sinks and sources can be listed with the commands:
pacmd list-sinks
pacmd list-sources
Pipewire implements both the PulseAudio and Jack APIs. It is therefore supported both via the Pulse and the Jack backends, and there is no need for a specific Pipewire backend.
Pipewire supports creating null-sink like PulseAudio. Create it with:
pactl load-module module-null-sink sink_name=MySink object.linger=1 media.class=Audio/Sink
List sources and sinks with:
pw-cli ls Node
This will list all devices, and the null-sink should be included like this:
id 75, type PipeWire:Interface:Node/3
factory.id = "18"
node.description = "MySink Audio/Sink sink"
node.name = "MySink"
media.class = "Audio/Sink"
This device can be set as the default output in the Gnome sound settings, meaning all desktop audio will use it. The audio sent to this device can then be captured from the monitor output named "MySink.monitor" using the PulseAudio backend.
Pipewire can also be configured to output to an ALSA Loopback. This is done by adding an ALSA sink in the Pipewire configuration. This sink then becomes available as an output device in the Gnome sound settings. See the "camilladsp-config" repository under Related projects for an example Pipewire configuration.
TODO test with Jack.
BlueALSA (bluez-alsa) is a project to receive or send audio through Bluetooth A2DP.
The Bluez source will connect to BlueALSA via D-Bus to get a file descriptor.
It will then read the audio directly from there, avoiding the need to go via ALSA.
Currently only capture (a2dp-sink) is supported.
BlueALSA is supported on Linux only, and requires building CamillaDSP with the bluez-backend Cargo feature.
Start by installing bluez-alsa.
Both Pipewire and PulseAudio will interfere with BlueALSA and must be disabled.
The source device should be paired after disabling Pipewire or PulseAudio and enabling BlueALSA.
Example configuration:
devices:
samplerate: 44100
chunksize: 4096
target_level: 8000
adjust_period: 3
resampler:
type: AsyncSinc
parameters:
type: Balanced
enable_rate_adjust: true
capture:
type: Bluez
format: S16LE
channels: 2
dbus_path: /org/bluealsa/hci0/dev_A0_B1_C2_D3_E4_F5/a2dpsnk/source
service: org.bluealsa (*)
After connecting an A2DP device, for example a mobile phone, the D-Bus path can be found with this command:
gdbus call -y --dest org.bluealsa -o /org/bluealsa -m org.freedesktop.DBus.ObjectManager.GetManagedObjects
This should produce output similar to this:
({objectpath '/org/bluealsa/hci0/dev_A0_B1_C2_D3_E4_F5/a2dpsnk/source': {'org.bluealsa.PCM1': {'Device': <objectpath '/org/bluez/hci0/dev_A0_B1_C2_D3_E4_F5'>, 'Sequence': <uint32 0>, 'Transport': <'A2DP-sink'>, 'Mode': <'source'>, 'Format': <uint16 33296>, 'Channels': <byte 0x02>, 'Sampling': <uint32 44100>, 'Codec': <'AAC'>, 'CodecConfiguration': <[byte 0x80, 0x01, 0x04, 0x03, 0x5b, 0x60]>, 'Delay': <uint16 150>, 'SoftVolume': <true>, 'Volume': <uint16 32639>}}},)
The wanted path is the string after objectpath.
If the output is looking like (@a{oa{sa{sv}}} {},), then no A2DP source is connected or detected.
Connect an A2DP device and try again. If a device is already connected, try removing and pairing the device again.
The service property can be left out to get the default. This only needs changing if there is more than one instance of BlueALSA running.
You have to specify correct capture sample rate, number of channel and sample format.
These parameters can be found with bluealsa-aplay:
> bluealsa-aplay -L
bluealsa:DEV=A0:B1:C2:D3:E4:F5,PROFILE=a2dp,SRV=org.bluealsa
MyPhone, trusted phone, capture
A2DP (AAC): S16_LE 2 channels 44100 Hz
Note that Bluetooth transfers data in chunks, and the time between chunks can vary. To avoid underruns, use a large chunksize and a large target_level. The values in the example above are a good starting point. Rate adjust should also be enabled.
CamillaDSP is using the YAML format for the configuration file. This is a standard format that was chosen because of its nice readable syntax. The Serde library is used for reading the configuration. There are a few things to keep in mind with YAML. The configuration is a tree, and the level is determined by the indentation level. For YAML the indentation is as important as opening and closing brackets in other formats. If it's wrong, Serde might not be able to give a good description of what the error is, only that the file is invalid. If you get strange errors, first check that the indentation is correct. Also check that you only use spaces and no tabs. Many text editors can help by highlighting syntax errors in the file.
The items at each level of the tree can be placed in any order. Consider the following example:
filters:
example_fir_a:
type: Conv
parameters:
filename: path/to/filter.txt <
format: TEXT <-- "filename", "format" and "type" can be in any order as long as they are properly indented to be part of the "parameters" block.
type: Raw <
example_fir_b:
parameters: <-- "parameters" can be placed before or after "type".
type: Wav
filename: path/to/filter.wav
type: Conv
mixers:
mono:
mapping:
- dest: 0
sources:
- channel: 0
gain: -6
- gain: -6 <-- The order of "gain" and "channel" can be reversed.
channel: 1 <
channels:
out: 1
in: 2
On the root level it contains filters and mixers. The mixers section could just as well be placed before the filters.
Looking at filters, the second filter swaps the order of parameters and type. Both variants are valid.
The mixer example shows that the gain and channel properties can be ordered freely.
There are two properties that are used to name and describe the configuration file. They are both optional.
title: "Example"
description: "Example description of a configuration"
Both these properties are optional and can be set to null or left out.
The title property is intended for a short title, while description can be longer and more descriptive.
There is a volume control that is enabled regardless of what configuration file is loaded.
CamillaDSP defines a total of five control "channels" for volume.
The default volume control reacts to the Main control channel.
When the volume or mute setting is changed, the gain is smoothly ramped to the new setting.
The duration of this ramp can be customized via the volume_ramp_time parameter
in the devices section.
The value must not be negative. If left out or set to null, it defaults to 400 ms.
In addition to this, there are four additional control channels, named Aux1 to Aux4.
These can be used to implement for example a separate volume control for a headphone output,
or crossfading between different input channels.
Example config:
devices:
samplerate: 96000
chunksize: 2048
queuelimit: 4 (*)
silence_threshold: -60 (*)
silence_timeout: 3.0 (*)
target_level: 500 (*)
adjust_period: 10 (*)
enable_rate_adjust: true (*)
resampler: null (*)
capture_samplerate: 44100 (*)
stop_on_rate_change: false (*)
rate_measure_interval: 1.0 (*)
volume_ramp_time: 400.0 (*)
capture:
type: Pulse
channels: 2
device: "MySink.monitor"
format: S16LE
playback:
type: Alsa
channels: 2
device: "hw:Generic_1"
format: S32LE
A parameter marked (*) in any example is optional. If they are left out from the configuration, or set to null, their default values will be used.
-
samplerateThe
sampleratesetting decides the sample rate that everything will run at. This rate must be supported by both the capture and playback device. -
chunksizeAll processing is done in chunks of data. The
chunksizeis the number of samples each chunk will have per channel. It's good if the number is an "easy" number like a power of two, since this speeds up the FFT in the Convolution filter. Suggested starting points for different sample rates:- 44.1 or 48 kHz: 1024
- 88.2 or 96 kHz: 2048
- 176.4 or 192 kHz: 4096
The duration in seconds of a chunk is
chunksize/samplerate, so the suggested values corresponds to about 22 ms per chunk. This is a reasonable value, and making it shorter can increase the cpu usage and make buffer underruns more likely.If you have long FIR filters you can reduce CPU usage by making the chunksize larger. When increasing, try increasing in factors of two, like 1024 -> 2048 or 4096 -> 8192.
-
queuelimit(optional, defaults to 4)The field
queuelimitshould normally be left out to use the default of 4. It sets the limit for the length of the queues between the capture device and the processing thread, as well as between the processing thread and the playback device. The total queue size limit will be2*chunksize*queuelimitsamples per channel.The value should only be changed if the capture device can provide data faster than the playback device can play it, like when using the Alsa "cdsp" plugin. If this case, set
queuelimitto a low value like 1. -
enable_rate_adjust(optional, defaults to false)This enables the playback device to control the rate of the capture device, in order to avoid buffer underruns or a slowly increasing latency. This is currently supported when using an Alsa, Wasapi or CoreAudio playback device (and any capture device). Setting the rate can be done in two ways.
- Some capture devices provide a way to adjust the speed of their virtual sample clock (also called pitch adjust). This is available with the Alsa Loopback and USB Audio gadget devices on Linux, as well as the latest (currently unreleased) version or BlackHole on macOS. When capturing from any of these devices, the adjustment can be done by tuning the virtual sample clock of the device. This avoids the need for asynchronous resampling.
- If asynchronous resampling is enabled, the adjustment can be done by tuning the resampling ratio.
Then
resamplermust be set to one of the "Async" types. This is supported for all capture devices.
With Alsa capture devices, the first option is used whenever it's available. If not, and when not using an Alsa capture device, then the second option is used.
-
target_level(optional, defaults to thechunksizevalue)The value is the number of samples that should be left in the buffer of the playback device when the next chunk arrives. Only applies when
enable_rate_adjustis set totrue. It will take some experimentation to find the right number. If it's too small there will be buffer underruns from time to time, and making it too large might lead to a longer input-output delay than what is acceptable. Suitable values are in the range 1/2 to 1 times thechunksize. -
adjust_period(optional, defaults to 10)The
adjust_periodparameter is used to set the interval between corrections, in seconds. The default is 10 seconds. Only applies whenenable_rate_adjustis set totrue. A smaller value will make for a faster reaction time, which may be useful if there are occasional buffer underruns when running with a smalltarget_levelto minimize latency. -
silence_threshold&silence_timeout(optional) The fieldssilence_thresholdandsilence_timeoutare optional and used to pause processing if the input is silent. The threshold is the threshold level in dB, and the level is calculated as the difference between the minimum and maximum sample values for all channels in the capture buffer. 0 dB is full level. Some experimentation might be needed to find the right threshold.The
silence_timeout(in seconds) is for how long the signal should be silent before pausing processing. Set this to zero, or leave it out, to never pause. -
resampler(optional, defaults tonull)Use this to configure a resampler. Setting it to
nullor leaving it out disables resampling . Configure a resampler to enable resampling of the input signal. In addition to resampling the input to a different sample rate, this can be useful for rate-matching capture and playback devices with independent clocks. See the Resampling section for more details. -
capture_samplerate(optional, defaults tonull)The capture samplerate. Setting it to
nullsets the capture samplerate to the same value assamplerate. If the resampler is only used for rate-matching, then the capture samplerate is the same as the overall samplerate, and this setting can be left out. -
stop_on_rate_changeandrate_measure_interval(both optional)Setting
stop_on_rate_changetotruemakes CamillaDSP stop the processing if the measured capture sample rate changes. Default isfalse. Therate_measure_intervalsetting is used for adjusting the measurement period. A longer period gives a more accurate measurement of the rate, at the cost of slower response when the rate changes. The default is 1.0 seconds. Processing will stop after 3 measurements in a row are more than 4% off from the configured rate. The value of 4% is chosen to allow some variation, while still catching changes between for example 44.1 to 48 kHz. -
volume_ramp_time(optional, defaults to 400 ms) This setting controls the duration of this ramp when changing volume of the default volume control. The value must not be negative. If left out or set tonull, it defaults to 400 ms. -
captureandplaybackInput and output devices are defined in the same way. A device needs:-
type: The available types depend on which features that were included when compiling. All possible types are:FileStdin(capture only)Stdout(playback only)Bluez(capture only)JackWasapiCoreAudioAlsaPulse
-
channels: number of channels -
device: device name (for Alsa, Pulse, Wasapi, CoreAudio). For CoreAudio and Wasapi, "default" will give the default device. -
filenamepath to the file (for File) -
format: sample format (for all except Jack).Currently supported sample formats are signed little-endian integers of 16, 24 and 32 bits as well as floats of 32 and 64 bits:
- S16LE - Signed 16-bit int, stored as two bytes
- S24LE - Signed 24-bit int, stored as four bytes (three bytes of data, one padding byte)
- S24LE3 - Signed 24-bit int, stored as three bytes (with no padding)
- S32LE - Signed 32-bit int, stored as four bytes
- FLOAT32LE - 32-bit float, stored as four bytes
- FLOAT64LE - 64-bit float, stored as eight bytes
Note that there are two 24-bit formats! Make sure to select the correct one.
Alsa Pulse Wasapi CoreAudio Jack File/Stdin/Stdout S16LE Yes Yes Yes Yes No Yes S24LE Yes Yes Yes Yes No Yes S24LE3 Yes Yes Yes Yes No Yes S32LE Yes Yes Yes Yes No Yes FLOAT32LE Yes Yes Yes Yes Yes Yes FLOAT64LE Yes No No No No Yes This table shows which formats in the different APIs are equivalent.
CamillaDSP Alsa Pulse S16LE S16_LE S16LE S24LE S24_LE S24_32LE S24LE3 S24_3LE S24LE S32LE S32_LE S32LE FLOAT32LE FLOAT_LE FLOAT32LE FLOAT64LE FLOAT64_LE -
The
Filedevice type reads or writes to a file, whileStdinreads from stdin andStdoutwrites to stdout. The format is raw interleaved samples, in the selected sample format. If the capture device reaches the end of a file, the program will exit once all chunks have been played. That delayed sound that would end up in a later chunk will be cut off. To avoid this, set the optional parameterextra_samplesfor the File capture device. This causes the capture device to yield the given number of samples (per channel) after reaching end of file, allowing any delayed sound to be played back. TheStdincapture device andStdoutplayback device use stdin and stdout, so it's possible to easily pipe audio between applications:> camilladsp stdio_capt.yml > rawfile.dat > cat rawfile.dat | camilladsp stdio_pb.ymlNote: On Unix-like systems it's also possible to use the File device and set the filename to
/dev/stdinfor capture, or/dev/stdoutfor playback.Please note the
Filecapture device isn't able to read wav-files directly. If you want to let CamillaDSP play wav-files, please see the separate guide for converting wav to raw files.Example config for File:
capture: type: File channels: 2 filename: "/path/to/inputfile.raw" format: S16LE extra_samples: 123 (*) skip_bytes: 0 (*) read_bytes: 0 (*) playback: type: File channels: 2 filename: "/path/to/outputfile.raw" format: S32LEExample config for Stdin/Stdout:
capture: type: Stdin channels: 2 format: S16LE extra_samples: 123 (*) skip_bytes: 0 (*) read_bytes: 0 (*) playback: type: Stdout channels: 2 format: S32LEThe
FileandStdincapture devices support two additional optional parameters, for advanced handling of raw files and testing:-
skip_bytes: Number of bytes to skip at the beginning of the file or stream. This can be used to skip over the header of some formats like .wav (which typically has a fixed size 44-byte header). Leaving it out or setting to zero means no bytes are skipped. -
read_bytes: Read only up until the specified number of bytes. Leave it out or set it to zero to read until the end of the file or stream. -
Example, this will skip the first 50 bytes of the file (index 0-49) and then read the following 200 bytes (index 50-249).
skip_bytes: 50 read_bytes: 200
See the separate readme for Wasapi.
See the separate readme for ALSA.
See the separate readme for CoreAudio.
The
Pulsecapture and playback devices have no advanced options.Example config for Pulse:
capture: type: Pulse channels: 2 device: "MySink.monitor" format: S16LE playback: type: Pulse channels: 2 device: "alsa_output.pci-0000_03_00.6.analog-stereo" format: S32LEThe
Jackcapture and playback devices do not have aformatparameter, since they always uses the FLOAT32LE format. It seems that thedeviceproperty should always be set to "default". This parameter may be removed in a future version.Example config for Jack:
capture: type: Jack channels: 2 device: "default" playback: type: Jack channels: 2 device: "default" -
Resampling is provided by the Rubato library.
This library does asynchronous and synchronous resampling with adjustable parameters. Asynchronous resampling can be done with or without anti-aliasing.
The resampler section under devices is used to specify the resampler.
Example:
resampler:
type: AsyncSinc
profile: Balanced
The resampler type is given by type, and the available options are:
- AsyncSinc
- AsyncPoly
- Synchronous
The types AsyncPoly and AsyncSinc need additional parameters, see each type below for details.
For asynchronous resampling with anti-aliasing, the overall strategy is to use a sinc interpolation filter with a fixed oversampling factor, and then use polynomial interpolation to get values for arbitrary times between those fixed points.
The AsyncSinc resampler takes an additional parameter profile.
This is used to select a pre-defined profile.
The Balanced profile is the best choice in most cases.
It provides good resampling quality with a noise threshold in the range
of -170 dB along with reasonable CPU usage.
As -170 dB is way beyond the resolution limit of even the best commercial DACs,
this preset is thus sufficient for all audio use.
The Fast and VeryFast profiles are faster but have a little more high-frequency roll-off
and give a bit higher resampling artefacts.
The Accurate profile provides the highest quality result,
with all resampling artefacts below -200dB, at the expense of higher CPU usage.
Example:
resampler:
type: AsyncSinc
profile: VeryFast
It is also possible to specify all parameters of the resampler instead of using the pre-defined profiles.
Example:
resampler:
type: AsyncSinc
sinc_len: 128
oversampling_factor: 256
interpolation: Cubic
window: Hann2
f_cutoff: null
Note that these two ways of defining the resampler cannot be mixed.
When using profile the other parameters must not be present and vice versa.
The f_cutoff parameter is the relative cutoff frequency of the anti-aliasing filter.
A value of 1.0 means the Nyquist limit. Useful values are in the range 0.9 - 0.99.
It can also be calculated automatically by setting f_cutoff to null.
Available interpolation types:
| Interpolation | Polynomial degree | Samples fitted |
|---|---|---|
| Nearest | 0 | 1 |
| Linear | 1 | 2 |
| Quadratic | 2 | 3 |
| Cubic | 3 | 4 |
See the Rubato documentation for a desciption of the other parameters.
For reference, the profiles are defined according to this table:
| VeryFast | Fast | Balanced | Accurate | |
|---|---|---|---|---|
| sinc_len | 64 | 128 | 192 | 256 |
| oversampling_factor | 1024 | 1024 | 512 | 256 |
| interpolation | Linear | Linear | Quadratic | Cubic |
| window | Hann2 | Blackman2 | BlackmanHarris2 | BlackmanHarris2 |
| f_cutoff | 0.91 (#) | 0.92 (#) | 0.93 (#) | 0.95 (#) |
(#) These cutoff values are approximate. The actual values used are calculated automatically at runtime for the combination of sinc length and window.
Asynchronous resampling without anti-aliasing works by performing polynomial interpolation between the sample points. This skips the computationally expensive sinc interpolation filter and is therefore considerably faster. This method produces a result that isn't as clean as with anti-aliasing, but the difference is small and often inaudible.
The polynomial interpolation uses the N nearest samples,
where the number of samples depends on the selected interpolation.
Example:
resampler:
type: AsyncPoly
interpolation: Cubic
Available interpolation types:
| Interpolation | Polynomial degree | Samples fitted |
|---|---|---|
| Linear | 1 | 2 |
| Cubic | 3 | 4 |
| Quintic | 5 | 6 |
| Septic | 7 | 8 |
Higher polynomial degrees produce higher quality results but use more processing power.
All are however considerably faster than the AsyncSinc type.
In theory these produce inferior quality compared to the AsyncSinc type with anti-aliasing,
however in practice the difference is small.
Use the AsyncPoly type to save processing power, with little or no perceived quality loss.
For performing fixed ratio resampling, like resampling
from 44.1kHz to 96kHz (which corresponds to a precise ratio of 147/320)
choose the Synchronous type.
This works by transforming the waveform with FFT, modifying the spectrum, and then getting the resampled waveform by inverse FFT.
This is considerably faster than the asynchronous variants, but does not support rate adjust.
The resampling quality is similar to the AsyncSinc type with the Accurate profile.
The Synchronous type takes no additional parameters:
resampler:
type: Synchronous
When using the rate adjust feature to match capture and playback devices,
one of the "Async" types must be used.
These asynchronous resamplers do not rely on a fixed resampling ratio.
When rate adjust is enabled the resampling ratio is dynamically adjusted in order to compensate
for drifts and mismatches between the input and output sample clocks.
Using the "Synchronous" type with rate adjust enabled will print warnings,
and any rate adjust request will be ignored.
A mixer is used to route audio between channels, and to increase or decrease the number of channels in the pipeline. Example for a mixer that copies two channels into four:
mixers:
ExampleMixer:
description: "Example mixer to convert two channels to four" (*)
channels:
in: 2
out: 4
mapping:
- dest: 0
mute: false (*)
sources:
- channel: 0
gain: 0 (*)
inverted: false (*)
scale: dB (*)
- dest: 1
mute: false (*)
sources:
- channel: 1
gain: 0 (*)
inverted: false (*)
scale: dB (*)
- dest: 2
sources:
- channel: 0
gain: 0 (*)
inverted: false (*)
scale: dB (*)
- dest: 3
sources:
- channel: 1
gain: 0 (*)
inverted: false (*)
scale: dB (*)
Parameters marked with (*) are optional. Set to null or leave out to use the default value.
The "channels" group define the number of input and output channels for the mixer. The mapping section then decides how to route the audio.
This is a list of the output channels, and for each channel there is a "sources" list that gives the sources for this particular channel.
Each source has a channel number, a gain value, a scale for the gain (dB or linear) and if it should be inverted (true/false).
A channel that has no sources will be filled with silence.
The mute option determines if an output channel of the mixer should be muted.
The mute, gain, scale and inverted parameters are optional, and defaults to not muted, a gain of 0 in dB, and not inverted.
The optional description property is intended for the user and is not used by CamillaDSP itself.
Another example, a simple stereo to mono mixer:
mixers:
mono:
channels:
in: 2
out: 1
mapping:
- dest: 0
sources:
- channel: 0
gain: -6
- channel: 1
gain: -6
Some audio interfaces bundle all their inputs together, meaning that it might be necessary to capture a large number of channels to get access to a particular input. To reduce the CPU load, CamillaDSP will try to avoid processing of any channel that is captured but not used in the pipeline.
Let's say we have an interface with one analog input, and one SPDIF. These are presented as a single 4-channel input where channels 0 and 1 are analog, 2 and 3 SPDIF. Then, setting the number of capture channels to 4 will enable both inputs. In this case we are only interested in the SPDIF input. This is then done by adding a mixer that reduces the number of channels to 2. In this mixer, input channels 0 and 1 are not mapped to anything. This is then detected, and no format conversion, resampling or processing will be done on these two channels.
The filters section defines the filter configurations to use in the pipeline. It's enough to define each filter once even if it should be applied on several channels. The supported filter types are Biquad, BiquadCombo and DiffEq for IIR and Conv for FIR. There are also filters just providing gain and delay. The last filter type is Dither, which is used to add dither when quantizing the output.
All filters take an optional description property.
This is intended for the user and is not used by CamillaDSP itself.
The gain filter simply changes the amplitude of the signal.
The inverted parameter simply inverts the signal.
This parameter is optional and the default is to not invert.
The gain value is given in either dB or as a linear factor, depending on the scale parameter.
This can be set to dB or linear.
If set to null or left out it defaults to dB.
When the dB scale is used (scale: dB), a positive gain value means the signal will be amplified while a negative values attenuates.
The gain value must be in the range -150 to +150 dB.
If linear gain is used (scale: linear), the gain value is treated as a simple multiplication factor.
A factor 0.5 attenuates by a factor two (equivalent to a gain of -6.02 dB).
A negative value inverts the signal.
Note that the invert setting also inverts, so a gain of -0.5 with invert set to true becomes inverted twice and the result is non-inverted.
The linear gain is limited to a range of -10.0 to +10.0.
The mute parameter determines if the the signal should be muted.
This is optional and defaults to not mute.
Example Gain filter:
filters:
gainexample_dB:
type: Gain
parameters:
gain: -6.0
inverted: false (*)
mute: false (*)
scale: dB (*)
gainexample_linear:
type: Gain
parameters:
gain: 0.5
inverted: false (*)
mute: false (*)
scale: linear (*)
The Volume filter is intended to be used as an additional volume control.
Note that the pipeline includes a volume control for the Main fader per default,
and it's not possible to select this fader for Volume filters.
Volume filters may use the four additional faders, named Aux1, Aux2,Aux3 and Aux4.
A Volume filter is configured to react to one of these faders.
The volume can then be changed via the websocket, by changing the corresponding fader.
A request to set the volume will be applied to all Volume filters listening to the affected fader.
When the volume or mute state is changed, the gain is ramped smoothly to the new value.
The duration of this ramp is set by the ramp_time parameter (unit milliseconds).
The value must not be negative. If left out or set to null, it defaults to 400 ms.
The value will be rounded to the nearest number of chunks.
Example Volume filter:
filters:
volumeexample:
type: Volume
parameters:
ramp_time: 200 (*)
fader: Aux1
The Loudness filter performs loudness compensation and is intended to be used in combination with a volume control.
Similar to a Volume filter, it reacts to the configured fader.
The available choices for fader are Main, Aux1, Aux2,Aux3 and Aux4.
Setting fader to Main adds loudness compensation to the default volume control.
By setting fader to one of the Aux faders it can instead work with a Volume filter
reacting to the same fader.
When used like this, there should only be a single Volume filter assigned to the chosed fader.
It can also be used with a volume control external to CamillaDSP.
The fader should then be set to one of the Aux faders, and the external volume control should update
this fader when the volume setting changes.
A special websocket command is provided for this, see the websocket command documentation.
If the external volume control is placed after CamillaDSP in the audio chain,
then the boost applied at high and low frequencies may cause clipping.
To avoid this, set attenuate_mid to true.
That makes the loudness filter attenuate the midband instead of boosting the extremes.
The method is the same as the one implemented by the RME ADI-2 DAC FS.
The loudness correction is done as shelving filters that boost the high (above 3500 Hz) and low (below 70 Hz) frequencies.
The amount of boost is adjustable with the high_boost and low_boost parameters. If left out, they default to 10 dB.
- When the volume is above the
reference_level, only gain is applied. - When the volume is below
reference_level- 20, the full correction is applied. - In the range between
reference_levelandreference_level-20, the boost value is scaled linearly.
In this figure, the reference_level was set to -5 dB, and high_boost = low_boost = 10 dB.
At a gain of 0 and -5, the curve is flat. Below that the boost increases.
At -15 dB half of the boost, and at -25 the full boost is applied.
Below -25 dB, the boost value stays constant.
Example Loudness filter, configured to work together with the default volume control:
filters:
loudness:
type: Loudness
parameters:
fader: Main (*)
reference_level: -25.0
high_boost: 7.0 (*)
low_boost: 7.0 (*)
attenuate_mid: false (*)
Allowed ranges:
- reference_level: -100 to +20
- high_boost: 0 to 20
- low_boost: 0 to 20
The delay filter provides a delay in milliseconds, millimetres or samples.
The unit can be ms, mm or samples, and if left out it defaults to ms.
When giving the delay in millimetres, the speed of sound of is assumed to be 343 m/s (dry air at 20 degrees Celsius).
If the subsample parameter is set to true, then it will use use an IIR filter to achieve subsample delay precision.
If set to false, the value will instead be rounded to the nearest number of full samples.
This is a little faster and should be used if subsample precision is not required.
The delay value must be positive or zero.
Example Delay filter:
filters:
delayexample:
type: Delay
parameters:
delay: 12.3
unit: ms
subsample: false
A FIR filter is given by an impulse response provided as a list of coefficients. The coefficients are preferably given in a separate file, but can be included directly in the config file. If the number of coefficients (or taps) is larger than the chunksize setting it will use segmented convolution. The number of segments is the filter length divided by the chunksize, rounded up.
Example FIR filters:
filters:
example_fir_a:
type: Conv
parameters:
type: Raw
filename: path/to/filter.txt
format: TEXT
skip_bytes_lines: 0 (*)
read_bytes_lines: 0 (*)
example_fir_b:
type: Conv
parameters:
type: Wav
filename: path/to/filter.wav
channel: 0 (*)
The type can be Raw, Wav or Values.
Use Wav to load a standard .wav file, Raw to load a raw file (see list of allowed raw formats below),
and Values for giving the coefficients directly in the configuration file.
The filename field should hold the path to the coefficient file.
Using the absolute path is recommended in most cases.
If a relative path is given it will first try to find the file relative to the config file path. If it's not found there, the path is assumed to be relative to the current working directory. Note that this only applies when the config is loaded from a file. When a config is supplied via the websocket server only the current working dir of the CamillaDSP process will be searched.
If the filename includes the tokens $samplerate$ or $channels$, these will be replaced by the corresponding values from the config.
For example, if samplerate is 44100, the filename /path/to/filter_$samplerate$.raw will be updated to /path/to/filter_44100.raw.
Example for giving values:
filters:
lowpass_fir:
type: Conv
parameters:
type: Values
values: [0.0, 0.1, 0.2, 0.3]
Setting the type to Dummy creates a dummy impulse response:
filters:
lowpass_fir:
type: Conv
parameters:
type: Dummy
length: 65536
This creates a dummy minumum-phase allpass filter of length length (that must be at least 1).
The first point has a value of one, and all the rest are zero: [1.0, 0.0, 0.0, ..., 0.0].
This is intended to provide an easy way to evaluate the CPU load for different filter lengths.
Supplying the coefficients as .wav file is the most convenient method.
The Wav type takes only one parameter channel.
This is used to select which channel of a multi-channel file to load.
For a standard stereo file, the left track is channel 0, and the right is channel 1.
This parameter is optional and defaults to 0 if left out.
The sample rate of the file is ignored.
To load coefficients from a raw file, use the Raw type. This is also used to load coefficients from text files.
Raw files are often saved with a .dbl, .raw, or .pcm ending.
The lack of a header means that the files doesn't contain any information about data format etc.
CamillaDSP supports loading coefficients from such files that contain a single channel only (stereo files are not supported), in all the most common sample formats.
The Raw type supports two additional optional parameters, for advanced handling of raw files and text files with headers:
skip_bytes_lines: Number of bytes (for raw files) or lines (for text) to skip at the beginning of the file. This can be used to skip over a header. Leaving it out or setting to zero means no bytes or lines are skipped.read_bytes_lines: Read only up until the specified number of bytes (for raw files) or lines (for text). Leave it out or set it to zero to read until the end of the file.
The filter coefficients can be provided either as text, or as raw samples. Each file can only hold one channel. The "format" parameter can be omitted, in which case it's assumed that the format is TEXT. This format is a simple text file with one value per row:
-0.000021
-0.000020
-0.000018
...
-0.000012
The other possible formats are raw data:
- S16LE: signed 16-bit little-endian integers
- S24LE: signed 24-bit little-endian integers stored as 32 bits (with the data in the low 24)
- S24LE3: signed 24-bit little-endian integers stored as 24 bits
- S32LE: signed 32-bit little-endian integers
- FLOAT32LE: 32-bit little endian float
- FLOAT64LE: 64-bit little endian float
IIR filters are implemented as Biquad filters. CamillaDSP can calculate the coefficients for a number of standard filters, or you can provide the coefficients directly.
Examples:
filters:
free_nbr1:
type: Biquad
parameters:
type: Free
a1: 1.0
a2: 1.0
b0: 1.0
b1: 1.0
b2: 1.0
hp_80:
type: Biquad
parameters:
type: Highpass
freq: 80
q: 0.5
peak_100:
type: Biquad
parameters:
type: Peaking
freq: 100
gain: -7.3
q: 0.5
peak_100_bw:
type: Biquad
parameters:
type: Peaking
freq: 100
gain: -7.3
bandwidth: 0.7
exampleshelf:
type: Biquad
parameters:
type: Highshelf
freq: 1000
gain: -12
slope: 6
exampleshelf_q:
type: Biquad
parameters:
type: Highshelf
freq: 1000
gain: -12
q: 1.5
LR_highpass:
type: BiquadCombo
parameters:
type: LinkwitzRileyHighpass
freq: 1000
order: 4
Single Biquads are defined using the type "Biquad". The available filter types are:
-
Free
Given by normalized coefficients
a1,a2,b0,b1,b2. -
Highpass & Lowpass
Second order high/lowpass filters (12dB/oct)
Defined by cutoff frequency
freqand Q-valueq. -
HighpassFO & LowpassFO
First order high/lowpass filters (6dB/oct)
Defined by cutoff frequency
freq. -
Highshelf & Lowshelf
High / Low uniformly affects the high / low frequencies respectively while leaving the low / high part unaffected. In between there is a slope of variable steepness.
Parameters:
freqis the center frequency of the sloping section.gaingives the gain of the filterslopeis the steepness in dB/octave. Values up to around +-12 are usable.qis the Q-value and can be used instead ofslopeto define the steepness of the filter. Only one ofqandslopecan be given.
-
HighshelfFO & LowshelfFO
First order (6dB/oct) versions of the shelving functions.
Parameters:
freqis the center frequency of the sloping section.gaingives the gain of the filter
-
Peaking
A parametric peaking filter with selectable gain
gainat a given frequencyfreqwith a bandwidth given either by the Q-valueqor bandwidth in octavesbandwidth. Note that bandwidth and Q-value are inversely related, a small bandwidth corresponds to a large Q-value etc. Use positive gain values to boost, and negative values to attenuate. -
Notch
A notch filter to attenuate a given frequency
freqwith a bandwidth given either by the Q-valueqor bandwidth in octavesbandwidth. The notch filter is similar to a Peaking filter configured with a large negative gain. -
GeneralNotch
The general notch is a notch where the pole and zero can be placed at different frequencies. It is defined by its zero frequency
freq_z, pole frequencyfreq_p, pole Qq_p, and an optional parameternormalize_at_dc.When pole and zero frequencies are different, the low-frequency gain is changed and the shape (peakiness) at the
freq_pside of the notch can be controlled byq_p. The response is similar to an adjustable Q 2nd order shelf betweenfreq_pandfreq_zplus a notch atfreq_z.The highpass-notch form is obtained when
freq_p>freq_z. In this form the LF (stopband) gain decreases to -X dB while the HF (passband) gain remains unchanged at 0 dB.The lowpass-notch form is obtained when
freq_p<freq_z. In this form, the LF (e.g. DC or passband) gain increases to X dB while the HF (e.g. stopband) gain remains at 0dB.The larger the difference between
freq_pandfreq_zthe larger is X.To automatically swap these levels, so that the LF gain remains at 0dB while the HF gain takes on the value of ±X dB, set the parameter
normalize_at_dctotrue(the default for this parameter isfalse). Note that when the pole and zero frequencies are set to the same value the common (symmetrical) notch is obtained. -
Bandpass
A second order bandpass filter for a given frequency
freqwith a bandwidth given either by the Q-valueqor bandwidth in octavesbandwidth. -
Allpass
A second order allpass filter for a given frequency
freqwith a steepness given either by the Q-valueqor bandwidth in octavesbandwidth -
AllpassFO
A first order allpass filter for a given frequency
freq. -
LinkwitzTransform
A normal sealed-box speaker has a second order high-pass frequency response given by a resonance frequency and a Q-value. A Linkwitz transform can be used to apply a tailored filter that modifies the actual frequency response to a new target response. The target is also a second order high-pass function, given by the target resonance frequency and Q-value.
Parameters:
freq_act: actual resonance frequency of the speaker.q_act: actual Q-value of the speaker.freq_target: target resonance frequency.q_target: target Q-value.
To build more complex filters, use the type "BiquadCombo". This automatically adds several Biquads to build other filter types. The available types are:
-
ButterworthHighpass & ButterworthLowpass
Defined by frequency,
freqand filterorder. -
LinkwitzRileyHighpass & LinkwitzRileyLowpass
Defined by frequency,
freqand filterorder.Note, the order must be even
-
Tilt
The "Tilt" filter applies a tilt across the entire audible spectrum. It takes a single parameter
gain. A positive value gives a positive tilt, that boosts the high end of the spectrum and attenuates the low. A negative value gives the opposite result.The
gainvalue is the difference in gain between the highest and lowest frequencies. It's applied symmetrically, so a value of +10 dB will result in 5 dB of boost at high frequencies, and 5 dB of attenuation at low frequencies. In between the gain changes linearly, with a midpoint at about 600 Hz.The
gainvalue is limited to +- 100 dB. -
FivePointPeq
This filter combo is mainly meant to be created by guis. Is defines a 5-point (or band) parametric equalizer by combining a Lowshelf, a Highshelf and three Peaking filters.
Each individual filter is defined by frequency, gain and q. The parameter names are:
- Lowshelf:
fls,gls,qls - Peaking 1:
fp1,gp1,qp1 - Peaking 2:
fp2,gp2,qp2 - Peaking 3:
fp3,gp3,qp3 - Highshelf:
fhs,ghs,qhs
All 15 parameters must be included in the config.
- Lowshelf:
Other types such as Bessel filters can be built by combining several Biquads. See the separate readme for more filter functions.
-
GraphicEqualizer
This creates a graphic equalizer with an arbitrary number of bands. It uses one Peaking biquad filter per band.
The range of the equalizer can be selected with the optional
freq_minandfreq_maxparameters. If left out, the range defaults to 20 Hz to 20 kHz.The number of bands, and the gain for each band is given by the
gainsparameter. This accepts a list of gains in dB. The number of values determines the number of bands. The gains are limited to +- 40 dB.The band frequencies are distributed evenly on the logarithmic frequency scale, and each band has the same relative bandwidth.
For example a 31-band equalizer on the default range gets a 1/3 octave bandwith, with the first three bands centered at 22.4, 27.9, 34.9 Hz, and the last two at 14.3 and 17.9 kHz.
Example:
filters: 5band_graphic: type: BiquadCombo parameters: type: GraphicEqualizer freq_min: 20 (*) freq_max: 20000 (*) gains: [0.0, 1.0, 2.0, 1.0, 0.0]The gain values are limited to the range +- 20 dB. Only the bands that have non-zero gain values are included in the processing, the ones with zero gain are skipped.
The "Dither" filter should only be added at the very end of the pipeline for each channel, and adds noise shaped dither to the output. This is intended for 16-bit output, but can be used also for higher bit depth if desired. There are several subtypes:
| Subtype | kHz | Noise shaping | Comments |
|---|---|---|---|
| None | Any | - | Bit depth reduction without dither, for testing purposes |
| Flat | Any | - | Triangular: objectively optimal non-shaped dither |
| Highpass | Any | 2 taps | Wannamaker highpassed, violet noise |
| Fweighted441 | 44.1 | 9 taps | Wannamaker, modeled after early ISO curve |
| - FweightedShort441 | 44.1 | 3 taps | - Lower cpu load, less noise but also less noise reduction |
| - FweightedLong441 | 44.1 | 24 taps | - Higher cpu load, less noise and nearly equal noise reduction |
| Gesemann441 | 44.1 | 8 taps | Modeled after LAME ATH curves with flattening |
| Gesemann48 | 48 | 8 taps | Modeled after LAME ATH curves with flattening |
| Lipshitz441 | 44.1 | 5 taps | Superseded by Fweighted441 |
| - LipshitzLong441 | 44.1 | 9 taps | - Superseded by FweightedLong441 |
| Shibata441 | 44.1 | 24 taps | Modeled after LAME ATH type 1 |
| - ShibataHigh441 | 44.1 | 20 taps | - High intensity (quieter noise) |
| - ShibataLow441 | 44.1 | 12 taps | - Low intensity (louder noise) |
| Shibata48 | 48 | 16 taps | Modeled after LAME ATH type 1 |
| - ShibataHigh48 | 48 | 28 taps | - High intensity (quieter noise) |
| - ShibataLow48 | 48 | 16 taps | - Low intensity (louder noise) |
| Shibata882 | 88.2 | 20 taps | Modeled after LAME ATH type 1 |
| - ShibataLow882 | 88.2 | 24 taps | - Low intensity (louder noise) |
| Shibata96 | 96 | 31 taps | Modeled after LAME ATH type 1 |
| - ShibataLow96 | 96 | 32 taps | - Low intensity (louder noise) |
| Shibata192 | 192 | 54 taps | Modeled after LAME ATH type 1 |
| - ShibataLow192 | 192 | 20 taps | - Low intensity (louder noise) |
The Shibata filters are the new filters from SSRC 1.32.
Filters with more taps are typically more precise, always at the expense of higher cpu load. Highpass is an exception, which is about as fast as Flat.
The parameter "bits" sets the target bit depth. For most oversampling delta-sigma DACs, this should match the bit depth of the playback device for best results. For true non-oversampling DACs, this should match the number of bits over which the DAC is linear (or the playback bit depth, whichever is lower). Setting it to a higher value is not useful since then the applied dither will be lost.
For the "Flat" subtype, the parameter "amplitude" sets the number of LSB to be dithered. To linearize the samples, this should be 2. Lower amplitudes produce less noise but also linearize less; higher numbers produce more noise but do not linearize more.
Some comparisons between the noise shapers are available from SoX, SSRC and ReSampler. To test the different types, set the target bit depth to something very small like 5 or 6 bits (the minimum allowed value is 2) and try them all. Note that on "None" this may well mean there is no or unintelligible audio -- this is to experiment with and show what the other ditherers actually do.
For sample rates above 48 kHz there is no need for anything more advanced than the "Highpass" subtype. For the low sample rates there is no spare bandwidth and the dither noise must use the audible range, with shaping to makes it less audible. But at 96 or 192 kHz there is all the bandwidth from 20 kHz up to 48 or 96 kHz where the noise can be placed without issues. The Highpass ditherer will place almost all of it there. Of course, the high-resolution Shibata filters provide some icing on the cake.
Selecting a noise shaping ditherer for a different sample rate than it was designed for, will cause the frequency response curve of the noise shaper to be fitted to the playback rate. This means that the curve no longer matches its design points to be minimally audible. You may experiment which shaper still sounds good, or use the Flat or Highpass subtypes which work well at any sample rate.
Example:
dither_fancy:
type: Dither
parameters:
type: Shibata441
bits: 16
The "Limiter" filter is used to limit the signal to a given level. It can use hard or soft clipping. Note that soft clipping introduces some harmonic distortion to the signal.
Example:
example_limiter:
type: Limiter
parameters:
soft_clip: false (*)
clip_limit: -10.0
Parameters:
soft_clip: enable soft clipping. Set tofalseto use hard clipping. Optional, defaults tofalse.clip_limit: the level in dB to clip at.
The "DiffEq" filter implements a generic difference equation filter with transfer function: H(z) = (b0 + b1z^-1 + .. + bnz^-n)/(a0 + a1z^-1 + .. + anz^-n). The coefficients are given as a list a0..an in that order. Example:
example_diffeq:
type: DiffEq
parameters:
a: [1.0, -0.1462978543780541, 0.005350765548905586]
b: [0.21476322779271284, 0.4295264555854257, 0.21476322779271284]
This example implements a Biquad lowpass, but for a Biquad the Free Biquad type is faster and should be preferred. Both a and b are optional. If left out, they default to [1.0].
The processors section contains the definitions for the Processors.
These are special "filters" that work on several channels at the same time.
At present only one type of processor, "Compressor", has been implemented.
Processors take an optional description property.
This is intended for the user and is not used by CamillaDSP itself.
The "Compressor" processor implements a standard dynamic range compressor. It is configured using the most common parameters.
Example:
processors:
democompressor:
type: Compressor
parameters:
channels: 2
attack: 0.025
release: 1.0
threshold: -25
factor: 5.0
makeup_gain: 15 (*)
clip_limit: 0.0 (*)
soft_clip: true (*)
monitor_channels: [0, 1] (*)
process_channels: [0, 1] (*)
pipeline:
- type: Processor
name: democompressor
Parameters:
channels: number of channels, must match the number of channels of the pipeline where the compressor is inserted.attack: time constant in seconds for attack, how fast the compressor reacts to an increase of the loudness.release: time constant in seconds for release, how fast the compressor scales back the compression when the loudness decreases.threshold: the loudness threshold in dB where compression sets in.factor: the compression factor, giving the amount of compression over the threshold. A factor of 4 means a sound that is 4 dB over the threshold will be attenuated to 1 dB over the threshold.makeup_gain: amount of fixed gain in dB to apply after compression. Optional, defaults to 0 dB.clip_limit: the level in dB to clip at. Providing a value enables clipping of the signal after compression. Leave out or set tonullto disable clipping.soft_clip: enable soft clipping. Set tofalseto use hard clipping. This setting is ignored when clipping is disabled. Note that soft clipping introduces some harmonic distortion to the signal. This setting is ignored ifenable_clip = false. Optional, defaults tofalse.monitor_channels: a list of channels used when estimating the loudness. Optional, defaults to all channels.process_channels: a list of channels that should be compressed. Optional, defaults to all channels.
The pipeline section defines the processing steps between input and output. The input and output devices are automatically added to the start and end. The pipeline section of the config is a list of processing steps. This determines both what processing steps that are applied, and in which order they are applied. A step can be a filter, a mixer or a processor. The filters, mixers and processors must be defined in the corresponding section of the configuration, and the pipeline refers to them by their name. During processing, the steps are applied in the listed order. For each mixer and for the output device the number of channels from the previous step must match the number of input channels. For filter steps, the channel number must exist at that point of the pipeline. Channels are numbered starting from zero. Apart from this, there are no rules for ordering of the steps or how many are added.
Each step take an optional description property. This is intended for the user and is not used by CamillaDSP itself.
Example:
pipeline:
- type: Mixer
description: "Expand to 4 channels"
name: to4channels
bypassed: false (*)
- type: Filter
description: "Left channel woofer"
channel: 0
bypassed: false (*)
names:
- lowpass_fir
- peak1
- type: Filter
description: "Right channel woofer"
channel: 1
bypassed: false (*)
names:
- lowpass_fir
- peak1
- type: Filter
description: "Left channel tweeter"
channel: 2
bypassed: false (*)
names:
- highpass_fir
- type: Filter
description: "Right channel tweeter"
channel: 3
bypassed: false (*)
names:
- highpass_fir
- type: Processor
description: "Compressor for protecting the drivers"
name: my_compressor
bypassed: false (*)
In this config first a mixer is used to copy a stereo input to four channels. Before the mixer, only channels 0 and 1 exist. After the mixer, four channels are available, with numbers 0 to 3. The mixer is followed by a filter step for each channel. Finally a compressor is added as the last step.
A filter step, type: Filter, can contain one or several filters.
The filters must be defined in the Filters section.
In the example above, channel 0 and 1 get filtered by lowpass_fir and peak1, while 2 and 3 get filtered by just highpass_fir.
If several filters are to be applied to a channel, it is recommended to put then in a single filter step.
This makes the config easier to overview and gives a minor performance benefit, compared to adding each filter in a separate step,
Mixer steps, type: Mixer, and processor steps, type: Processor, are defined in a similar way.
These steps take just the the name of a mixer of processor defined in the Mixers or Processors section.
If the name of a mixer, processor or filter includes the tokens $samplerate$ or $channels$,
these will be replaced by the corresponding values from the config.
For example, if the samplerate is 44100, the filter name fir_$samplerate$ will be updated to fir_44100.
Each pipeline step has an optional bypassed property. Setting this to true removes this step from the pipeline.
Take care when bypassing mixers. If a mixer is used to change the number of channels (like the one in the example above),
then bypassing it will make the pipeline output the wrong number of channels.
In this case, the bypass may be used to switch between mixers with different settings.
REW can automatically generate a set of filters for correcting the frequency response of a system. REW V5.20.14 and later is able to export the filters in the CamillaDSP yaml format.
- Go to the "EQ Filters" screen. Expand the "Equalizer" section in the list on the right side.
- Select "CamillaDSP" as Manufacturer and "Filters" as Model.
- Expand the "Filter Task" section and click "Save filter settings to YAML file".
- This opens a popup with the the text "Enter the label to use for each filter, the filter number will be appended to the label". This allows identification of the filter set.
Note that the generated YAML file is not a complete CamillaDSP configuration. It contains only filter definitions and pipeline steps, that can be pasted into a CamillaDSP config file.
Please note that the show_config.py script mentioned here is deprecated, and has been replaced by the plotcamillaconf tool from the pycamilladsp-plot library.
The new tool provides the same functionality as well as many improvements.
The show_config.py does not support any of the newer config options, and the script will be removed in a future version.
A Python script is included to view the configuration. This plots the transfer functions of all included filters, as well as plots a flowchart of the entire processing pipeline. Run it with:
python show_config.py /path/to/config.yml
Example flowchart:
Note that the script assumes a valid configuration file and will not give any helpful error messages if it's not, so it's a good idea to first use CamillaDSP to validate the file. The script requires the following:
- Python 3
- Numpy
- Matplotlib
- PyYAML
Other projects using CamillaDSP:
- https://github.com/scripple/alsa_cdsp - ALSA CamillaDSP "I/O" plugin, automatic config updates at changes of samplerate, sample format or number of channels.
- https://github.com/raptorlightning/I2S-Hat - An SPDIF Hat for the Raspberry Pi 2-X for SPDIF Communication, see also this thread at diyAudio.com.
- https://github.com/daverz/camilla-remote-control - Interface for remote control of CamillaDSP using a FLIRC USB infrared receiver or remote keyboard.
- https://github.com/Wang-Yue/CamillaDSP-Monitor - A script that provides a DSP pipeline and a spectral analyzer similar to those of the RME ADI-2 DAC/Pro.
Music players:
- https://moodeaudio.org/ - moOde audio player, audiophile-quality music playback for Raspberry Pi.
- https://github.com/thoelf/Linux-Stream-Player - Play local files or streamed music with room EQ on Linux.
- https://github.com/Lykkedk/SuperPlayer-v8.0.0---SamplerateChanger-v1.0.0 - Automatic filter switching at sample rate change for squeezelite, see also this thread at diyAudio.com.
Guides and example configs:
- https://github.com/ynot123/CamillaDSP-Cfgs-FIR-Examples - Example Filter Configuration and Convolver Coefficients.
- https://github.com/hughpyle/raspot - Hugh's raspotify config
- https://github.com/Wang-Yue/camilladsp-crossfeed - Bauer stereophonic-to-binaural crossfeed for headphones
- https://github.com/jensgk/akg_k702_camilladsp_eq - Headphone EQ and Crossfeed for the AKG K702 headphones
- https://github.com/phelluy/room_eq_mac_m1 - Room Equalization HowTo with REW and Apple Silicon
Projects of general nature which can be useful together with CamillaDSP:
- https://github.com/scripple/alsa_hook_hwparams - Alsa hooks for reacting to sample rate and format changes.
- https://github.com/HEnquist/cpal-listdevices - List audio devices with names and supported formats under Windows and macOS.
See the list of frequently asked questions.
See the trouble troubleshooting guide for explanations of most error messages.