Skip to content


Switch branches/tags

Name already in use

A tag already exists with the provided branch name. Many Git commands accept both tag and branch names, so creating this branch may cause unexpected behavior. Are you sure you want to create this branch?


Failed to load latest commit information.
Latest commit message
Commit time
April 23, 2021 22:51
January 12, 2022 22:27
October 25, 2022 23:59
April 25, 2021 00:05
July 6, 2020 20:28
May 3, 2020 16:41
December 27, 2020 20:45
December 27, 2020 20:41
December 27, 2020 20:41
February 6, 2020 23:32
November 10, 2022 20:50
December 11, 2019 23:13
March 1, 2020 15:32
August 3, 2021 22:02
January 20, 2021 16:08
February 12, 2020 08:48
June 4, 2021 22:40

CamillaDSP v1.0

CI test and lint

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.

Table of Contents




How to run

Processing audio


Related projects

Getting help



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.

How it works

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.

Websocket server

The websocket server launches a separate thread to handle each connected client. All commands to change the config are sent to the supervisor thread.

System requirements

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%.

Linux requirements

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)

Windows requirements

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.

MacOS requirements

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

Usage example: crossover for 2-way speakers

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.

Companion libraries and tools

These projects are part of the CamillaDSP family:


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 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.gz


Use 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:

For Windows you also need the "Build Tools for Visual Studio". Get them from here:

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.

Building in Linux with standard features

  • Install pkg-config (very likely already installed):
    • Fedora: sudo dnf install pkgconf-pkg-config
    • Debian/Ubuntu etc: sudo apt-get install pkg-config
    • Arch: sudo pacman -S cargo pkg-config
  • Install Alsa dependency:
    • Fedora: sudo dnf install alsa-lib-devel
    • Debian/Ubuntu etc: sudo apt-get install libasound2-dev
    • Arch: sudo pacman -S alsa-lib
  • Install OpenSSL dependency:
    • Fedora: sudo dnf install openssl openssl-devel
    • Debian/Ubuntu etc: sudo apt-get install openssl libssl-dev
    • Arch: sudo pacman -S openssl
  • 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 install command takes the same options for features as the build command.

Customized build

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.
  • websocket: Websocket server for control
  • secure-websocket: Enable secure websocket, also enables the websocket feature
  • FFTW: Use FFTW instead of RustFFT
  • 32bit: Perform all calculations with 32-bit floats (instead of 64)
  • neon: Enable the experimental Neon support for aarch64 in the resampler. Note that this only works on 64-bit arm, and requires a very recent nightly rust compiler.

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.

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

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
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
cargo install --path . --no-default-features --features FFTW --features 32bit

Optimize for your system

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 

Building on Windows and macOS

The platform-specific backends are always enabled when building on Windows and macOS. The recommended build command is:


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.

How to run

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.

Command line options

Starting with the --help flag prints a short help message:

> camilladsp.exe --help
CamillaDSP 0.6.1
Henrik Enquist <>
A flexible tool for processing audio

Built with features: websocket

Supported device types:
Capture: File, Stdin, Wasapi
Playback: File, Stdout, Wasapi

    camilladsp.exe [FLAGS] [OPTIONS] <configfile>

    -m, --mute       Start with Volume and Loudness filters 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

    -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 Volume and Loudness filters
    -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]

    <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.


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 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 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.

Overriding config values

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.

Volume control

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):

-g 12.3
--gain 12.3


These will NOT work:

-g -12.3
--gain -12.3

Exit codes

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

Reloading the configuration

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.

Controlling via websocket

See the separate readme for the websocket server

Processing audio

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.

File or pipe

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.

MacOS (CoreAudio)

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 = "18"
 		node.description = "MySink Audio/Sink sink" = "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.


The YAML format

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:

    type: Conv
      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                     <
    parameters:                     <-- "parameters" can be placed before or after "type".
      type: Wav 
      filename: path/to/filter.wav
    type: Conv

      - dest: 0
          - channel: 0
            gain: -6
          - gain: -6                <-- The order of "gain" and "channel" can be reversed.
            channel: 1              <
      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.


Example config:

  samplerate: 96000
  chunksize: 1024
  queuelimit: 4 (*)
  silence_threshold: -60 (*)
  silence_timeout: 3.0 (*)
  target_level: 500 (*)
  adjust_period: 10 (*)
  enable_rate_adjust: true (*)
  enable_resampling: true (*)
  resampler_type: BalancedAsync (*)
  capture_samplerate: 44100 (*)
  stop_on_rate_change: false (*)
  rate_measure_interval: 1.0 (*)
    type: Pulse
    channels: 2
    device: "MySink.monitor"
    format: S16LE
    type: Alsa
    channels: 2
    device: "hw:Generic_1"
    format: S32LE

Any parameter marked (*) in all examples in this section are optional. If they are left out from the configuration, their default values will be used.

  • samplerate

    The samplerate setting decides the sample rate that everything will run at. This rate must be supported by both the capture and playback device.

  • chunksize

    All processing is done in chunks of data. The chunksize is 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 queuelimit should 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, and between the processing thread and the playback device. The total queue size limit will be 2*chunksize*queuelimit samples per channel. The maximum RAM usage is 8*2*chunksize*queuelimit bytes. For example at the default setting of 128 and a chunksize of 1024, the total size limit of the queues is about 2MB (or 1MB if the 32bit compile option is used). The queues are allocated as needed, this value only sets an upper limit.

    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 queuelimit to 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 of a slowly increasing latency. This is currently supported when using an Alsa, Wasapi or CoreAudio playback device. Setting the rate can be done in two ways.

    • If the capture device is an Alsa Loopback device, the adjustment is done by tuning the virtual sample clock of the Loopback device. This avoids any need for resampling.
    • If resampling is enabled, the adjustment is done by tuning the resampling ratio. The resampler_type must then be one of the "Async" variants.
  • target_level (optional, defaults to the chunksize value)

    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_adjust is set to true. 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 the chunksize.

  • adjust_period (optional, defaults to 10)

    The adjust_period parameter is used to set the interval between corrections, in seconds. The default is 10 seconds. Only applies when enable_rate_adjust is set to true. A smaller value will make for a faster reaction time, which may be useful if there are occasional buffer underruns when running with a small target_level to minimize latency.

  • silence_threshold & silence_timeout (optional) The fields silence_threshold and silence_timeout are 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.

  • enable_resampling (optional, defaults to false)

    Set this to true 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.

  • resampler_type (optional, defaults to "BalancedAsync")

    The resampler type to use. Valid choices are "Synchronous", "FastAsync", "BalancedAsync", "AccurateAsync", "FreeAsync".

    If used for rate matching with enable_rate_adjust: true the one of the "Async" variants must be used. See also the Resampling section.

  • capture_samplerate (optional, defaults to value of samplerate)

    The capture samplerate. 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_change and rate_measure_interval (both optional)

    Setting stop_on_rate_change to true makes CamillaDSP stop the processing if the measured capture sample rate changes. Default is false. The rate_measure_interval setting 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.

  • capture and playback Input 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:

      • File
      • Stdin (capture only)
      • Stdout (playback only)
      • Jack
      • Wasapi
      • CoreAudio
      • Alsa
      • Pulse
    • channels: number of channels

    • device: device name (for Alsa, Pulse, Wasapi, CoreAudio). For CoreAudio and Wasapi, "default" will give the default device.

    • filename path 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.

      Supported formats

      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

      Equivalent formats

      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
      FLOAT64LE FLOAT64_LE -

    File, Stdin, Stdout

    The File device type reads or writes to a file, while Stdin reads from stdin and Stdout writes 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 parameter extra_samples for 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. The Stdin capture device and Stdout playback 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.yml

    Note: On Unix-like systems it's also possible to use the File device and set the filename to /dev/stdin for capture, or /dev/stdout for playback.

    Please note the File capture 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:

        type: File
        channels: 2
        filename: "/path/to/inputfile.raw"
        format: S16LE
        extra_samples: 123 (*)
        skip_bytes: 0 (*)
        read_bytes: 0 (*)
        type: File
        channels: 2
        filename: "/path/to/outputfile.raw"
        format: S32LE

    Example config for Stdin/Stdout:

        type: Stdin
        channels: 2
        format: S16LE
        extra_samples: 123 (*)
        skip_bytes: 0 (*)
        read_bytes: 0 (*)
        type: Stdout
        channels: 2
        format: S32LE

    The File and Stdin capture 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 Pulse capture and playback devices have no advanced options.

    Example config for Pulse:

        type: Pulse
        channels: 2
        device: "MySink.monitor"
        format: S16LE
        type: Pulse
        channels: 2
        device: "alsa_output.pci-0000_03_00.6.analog-stereo"
        format: S32LE


    The Jack capture and playback devices do not have a format parameter, since they always uses the FLOAT32LE format. It seems that the device property should always be set to "default". This parameter may be removed in a future version.

    Example config for Jack:

        type: Jack
        channels: 2
        device: "default"
        type: Jack
        channels: 2
        device: "default"


Resampling is provided by the Rubato library.

This library does asynchronous and synchronous resampling with adjustable parameters. For asynchronous resampling, the overall strategy is to use a sinc interpolation filter with a fixed oversampling ratio, and then use polynomial interpolation to get values for arbitrary times between those fixed points. For synchronous resampling it instead works by transforming the waveform with FFT, modifying the spectrum, and then getting the resampled waveform by inverse FFT.

CamillaDSP provides four preset profiles for the resampler:

  • Synchronous
  • FastAsync
  • BalancedAsync
  • AccurateAsync

The "BalancedAsync" preset is the best choice in most cases, if an asynchronous resampler is needed. It provides good resampling quality with a noise threshold in the range of -150 dB along with reasonable CPU usage. As -150 dB is way beyond the resolution limit of even the best commercial DACs, this preset is thus sufficient for all audio use. The "FastAsync" preset is faster but have a little more high-frequency roll-off and give a bit higher resampling artefacts. The "AccurateAsync" preset provide the highest quality result, with all resampling artefacts below -200dB, at the expense of higher CPU usage. There is also a "FreeAsync" mode as well where all parameters can be set freely. The configuration is specified like this:

      f_cutoff: 0.9
      sinc_len: 128
      window: Hann2
      oversampling_ratio: 128
      interpolation: Cubic

For reference, the asynchronous presets are defined according to this table:

FastAsync BalancedAsync AccurateAsync
sinc_len 64 128 256
oversampling_ratio 1024 1024 256
interpolation Linear Linear Cubic
window Hann2 Blackman2 BlackmanHarris2
f_cutoff 0.915 0.925 0.947

For performing fixed ratio resampling, like resampling from 44.1kHz to 96kHz (which corresponds to a precise ratio of 147/320) choose the "Synchronous" variant. This is considerably faster than the asynchronous variants, but does not support rate adjust. The quality is comparable to the "AccurateAsync" preset.

When using the rate adjust feature to match capture and playback devices, one of the "Async" variants must be used. These asynchronous presets 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" variant with rate adjust enabled will print warnings, and any rate adjust request will be ignored.

See the library documentation for more details. Rubato on


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:

      in: 2
      out: 4
      - dest: 0
        mute: false (*)
          - channel: 0
            gain: 0 (*)
            inverted: false (*)
      - dest: 1
        mute: false (*)
          - channel: 1
            gain: 0 (*)
            inverted: false (*)
      - dest: 2
          - channel: 0
            gain: 0 (*)
            inverted: false (*)
      - dest: 3
          - channel: 1
            gain: 0 (*)
            inverted: false (*)

Parameters marked with (*) are optional. 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 in dB, 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 and inverted parameters are optional, and defaults to not muted, a gain of 0 dB, and not inverted.

Another example, a simple stereo to mono mixer:

      in: 2
      out: 1
      - dest: 0
          - channel: 0
            gain: -6
          - channel: 1
            gain: -6

Skip processing of unused channels

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.


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 dB, and a positive value means the signal will be amplified while a negative values attenuates. The gain value must be in the range -150 to +150 dB. The mute parameter determines if the the signal should be muted. This is optional and defaults to not mute.

Example Gain filter:

    type: Gain
      gain: -6.0 
      inverted: false
      mute: false (*)


The Volume filter is intended to be used as a volume control. The initial volume and muting state can be set with the gain and mute command line parameters. The volume can then be changed via the websocket. A request to set the volume will be applied to all Volume filters. 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, it defaults to 200 ms. The value will be rounded to the nearest number of chunks. To use this filter, insert a Volume filter somewhere in the pipeline for each channel. It's possible to use this to make a dithered volume control by placing the Volume filter somewhere in the pipeline, and having a Dither filter as the last step.

Example Volume filter:

    type: Volume
      ramp_time: 200


The Loudness filter is intended to be used as a volume control, similarly to the Volume filter. See the Volume filter for a description of how it is used. The difference is that the Loudness filter applies loudness correction when the volume is lowered. 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_level and reference_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:

    type: Loudness
      ramp_time: 200.0
      reference_level: -25.0 
      high_boost: 7.0
      low_boost: 7.0

Allowed ranges:

  • reference_level: -100 to 0
  • 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:

    type: Delay
      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:

    type: Conv
      type: Raw 
      filename: path/to/filter.txt
      format: TEXT
      skip_bytes_lines: 0 (*)
      read_bytes_lines: 0 (*)
    type: Conv
      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.

Values directly in config file

Example for giving values:

    type: Conv
      type: Values
      values: [0.0, 0.1, 0.2, 0.3]
      length: 12345

The length setting is optional. It is used to extend the number of coefficients past the ones given in values. The added coefficients are all zeroes. This is intended to provide an easy way to evaluating the CPU load for different filter lengths.

For testing purposes the entire "parameters" block can be left out (or commented out with a # at the start of each line). This then becomes a dummy filter that does not affect the signal.

Coefficients from Wav-file

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.

Coefficient Raw (headerless) data file

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:


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.


    type: Biquad
      type: Free
      a1: 1.0
      a2: 1.0
      b0: 1.0
      b1: 1.0
      b2: 1.0
    type: Biquad
      type: Highpass
      freq: 80
      q: 0.5
    type: Biquad
      type: Peaking
      freq: 100
      gain: -7.3
      q: 0.5
    type: Biquad
      type: Peaking
      freq: 100
      gain: -7.3
      bandwidth: 0.7
    type: Biquad
      type: Highshelf
      freq: 1000
      gain: -12
      slope: 6
    type: Biquad
      type: Highshelf
      freq: 1000
      gain: -12
      q: 1.5
    type: BiquadCombo
      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 freq and Q-value q.

  • 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.


    • freq is the center frequency of the sloping section.
    • gain gives the gain of the filter
    • slope is the steepness in dB/octave. Values up to around +-12 are usable.
    • q is the Q-value and can be used instead of slope to define the steepness of the filter. Only one of q and slope can be given.
  • HighshelfFO & LowshelfFO

    First order (6dB/oct) versions of the shelving functions.


    • freq is the center frequency of the sloping section.
    • gain gives the gain of the filter
  • Peaking

    A parametric peaking filter with selectable gain gain at a given frequency freq with a bandwidth given either by the Q-value q or bandwidth in octaves bandwidth. 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 freq with a bandwidth given either by the Q-value q or bandwidth in octaves bandwidth. The notch filter is similar to a Peaking filter configured with a large negative gain.

  • Bandpass

    A second order bandpass filter for a given frequency freq with a bandwidth given either by the Q-value q or bandwidth in octaves bandwidth.

  • Allpass

    A second order allpass filter for a given frequency freq with a steepness given either by the Q-value q or bandwidth in octaves bandwidth

  • 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.


    • 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, freq and filter order.

  • LinkwitzRileyHighpass & LinkwitzRileyLowpass

    Defined by frequency, freq and filter order.

    Note, the order must be even

  • 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.

Other types such as Bessel filters can be built by combining several Biquads. See the separate readme for more filter functions.


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 types, and the parameter "bits" sets the target bit depth. For the best result this should match the bit depth of the playback device. Setting it to a higher value is not useful since then the applied dither will be rounded off. On the other hand, setting it to a much lower value, for example 5 or 6 bits (minimum allowed value is 2), makes the noise very audible and can be useful for comparing the different types.


    type: Dither
      type: Lipshitz441
      bits: 16

The available types are

  • Simple, simple noise shaping with increasing noise towards higher frequencies
  • Uniform, just dither, no shaping. Requires also the parameter "amplitude" to set the dither amplitude in units of LSB (least significant bit). The allowed amplitude range is 0 to 100.
  • Lipshitz441, for 44.1 kHz
  • Fweighted441, for 44.1 kHz
  • Shibata441, for 44.1 kHz
  • Shibata48, for 48 kHz
  • ShibataLow441, for 44.1 kHz
  • ShibataLow48, for 48 kHz
  • None, just quantize without dither. Only useful with small target bit depth for demonstration.

Lipshitz, Fweighted and Shibata give the least amount of audible noise. See the SOX documentation for more details. To test the different types, set the target bit depth to something very small like 5 bits and try them all.

For sample rates above 48 kHz there is no need for anything more advanced than the "Simple" type. 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 20kHz up to 48 or 96kHz where the noise can be placed without issues. The Simple type will place almost all of it there.

Difference equation

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 in that order. Example:

    type: DiffEq
      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 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 is essentially a list of filters and/or mixers. There are no rules for ordering or how many are added. For each mixer and for the output device the number of channels from the previous step must match the number of input channels.


  - type: Mixer
    name: to4channels
  - type: Filter
    channel: 0
      - lowpass_fir
      - peak1
  - type: Filter
    channel: 1
      - lowpass_fir
      - peak1
  - type: Filter
    channel: 2
      - highpass_fir
  - type: Filter
    channel: 3
      - highpass_fir

In this config first a mixer is used to copy a stereo input to four channels. Then for each channel a filter step is added. A filter block can contain one or several filters that must be define in the "Filters" section. Here channel 0 and 1 get filtered by "lowpass_fir" and "peak1", while 2 and 3 get filtered by just "highpass_fir". If the names of mixers or filters includes the tokens $samplerate$ or $channels$, these will be replaced by the corresponding values from the config. For example, if samplerate is 44100, the filter name fir_$samplerate$ will be updated to fir_44100.

Translating filters exported by REW

REW can automatically generate a set of filters for correcting the response. These can then be exported as an .xml-file. This file can then be translated to CamillaDSP filters using the Python script. This will generate filters and pipeline steps that can be pasted into a CamillaDSP config file. This script currently supports only Peaking filters.

Visualizing the config

Please note that the 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 does not support any of 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 /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

Related projects

Other projects using CamillaDSP:

Music players:

Guides and example configs:

Projects of general nature which can be useful together with CamillaDSP:

Getting help


See the list of frequently asked questions.


See the trouble troubleshooting guide for explanations of most error messages.