About

dsp is an audio processing program with an interactive mode.

Building

Dependencies

• GNU Make
• pkg-config

Optional dependencies

• fftw3: For matrix4_mb, resample, fir, fir_p, and hilbert effects.
• zita-convolver: For the zita_convolver effect.
• libsndfile: For sndfile input/output support (recommended).
• ffmpeg (libavcodec, libavformat, and libavutil): For ffmpeg input support.
• alsa-lib: For alsa input/output support.
• libao: For ao output support.
• libmad: For mp3 input support (disabled by default).
• libpulse-simple: For PulseAudio input/ouput support.
• LADSPA: For the LADSPA frontend and the ladspa_host effect.

Build

$ make

Run ./configure [options] manually if you want to build with non-default options. Run ./configure --help to see all available options.

Install

# make install

Synopsis

dsp [options] path ... [effect [args]] ...

Options

Global options

Option	Description
-h	Show help text.
-b frames	Block size (must be given before the first input).
-i	Force interactive mode.
-I	Disable interactive mode.
-q	Disable progress display.
-s	Silent mode.
-v	Verbose mode.
-d	Force dithering.
-D	Disable dithering.
-E	Don't drain effects chain before rebuilding.
-p	Plot effects chain magnitude response instead of processing audio.
-P	Same as -p, but also plot phase response.
-V	Verbose progress display.
-S	Use "sequence" input combining mode.

Input/output options

Option	Description
-o	Output.
-t type	Type.
-e encoding	Encoding.
-B/L/N	Big/little/native endian.
-r frequency[k]	Sample rate.
-c channels	Number of channels.
-R ratio	Buffer ratio.
-n	Equivalent to -t null null.

Inputs and Outputs

Supported input/output types

Type	Modes	Encodings
null	rw	sample_t
sgen	r	sample_t
sndfile	r	autodetected
wav	rw	s16 u8 s24 s32 float double mu-law a-law ima_adpcm ms_adpcm gsm6.10 nms_adpcm_16 nms_adpcm_24 nms_adpcm_32 g721_32 mpeg2.3
aiff	rw	s16 s8 u8 s24 s32 float double mu-law a-law ima_adpcm gsm6.10 dwvw_12 dwvw_16 dwvw_24
au	rw	s16 s8 s24 s32 float double mu-law a-law g721_32 g723_24 g723_40
raw	rw	s16 s8 u8 s24 s32 float double mu-law a-law gsm6.10 vox_adpcm nms_adpcm_16 nms_adpcm_24 nms_adpcm_32 dwvw_12 dwvw_16 dwvw_24
paf	rw	s16 s8 s24
svx	rw	s16 s8
nist	rw	s16 s8 s24 s32 mu-law a-law
voc	rw	s16 u8 mu-law a-law
ircam	rw	s16 s32 float mu-law a-law
w64	rw	s16 u8 s24 s32 float double mu-law a-law ima_adpcm ms_adpcm gsm6.10
mat4	rw	s16 s32 float double
mat5	rw	s16 u8 s32 float double
pvf	rw	s16 s8 s32
xi	rw	dpcm_8 dpcm_16
htk	rw	s16
sds	rw	s16 s8 s24
avr	rw	s16 s8 u8
wavex	rw	s16 u8 s24 s32 float double mu-law a-law
sd2	rw	s16 s8 s24 s32
flac	rw	s16 s8 s24
caf	rw	s16 s8 s24 s32 float double mu-law a-law alac_16 alac_20 alac_24 alac_32
wve	rw	a-law
ogg	rw	vorbis opus
mpc2k	rw	s16
rf64	rw	s16 u8 s24 s32 float double mu-law a-law
sf/mpeg	rw	mpeg1.1 mpeg1.2 mpeg2.3
ffmpeg	r	autodetected
alsa	rw	s16 u8 s8 s24 s24_3 s32 float double
ao	w	s16 u8 s32
mp3	r	mad_f
pcm	rw	s16 u8 s8 s24 s24_3 s32 float double
pulse	rw	s16 u8 s24 s24_3 s32 float

Input combining modes

In concatenate mode (the default), the inputs are concatenated in the order given and sent to the output. All inputs must have the same sample rate and number of channels.

In sequence mode, the inputs are sent serially to the output like concatenate mode, but the inputs do not need to have the same sample rate or number of channels. The effects chain and/or output will be rebuilt/reopened when required. Note that if the output is a file, the file will be truncated if it is reopened. This mode is most useful when the output is an audio device, but can also be used to concatenate inputs with different sample rates and/or numbers of channels into a single output file when used with the resample and/or remix effects.

Signal generator

The sgen input type is a basic (for now, at least) signal generator that can generate impulses and exponential sine sweeps. The syntax for the path argument is as follows:

[type[@channel_selector][:arg[=value]...]][/type...][+len[s|m|S]]

type may be sine for sine sweeps or tones, or delta for a delta function (impulse). sine accepts the following arguments:

• freq=f0[k][-f1[k]] Frequency. If len is set and f1 is given, an exponential sine sweep is generated.

The arguments for delta are:

• offset=time[s|m|S] Offset in seconds, miliseconds or samples.

Example:

$ dsp -t sgen -c 2 sine@0:freq=500-1k/sine@1:freq=300-800+2 gain -10

Effects

Complete effects list

• lowpass_1 f0[k]
  First-order lowpass filter.

• highpass_1 f0[k]
  First-order highpass filter.

• allpass_1 f0[k]
  First-order allpass filter.

• lowshelf_1 f0[k] gain
  First-order lowshelf filter.

• highshelf_1 f0[k] gain
  First-order highshelf filter.

• lowpass_1p f0[k]
  Single pole lowpass (EWMA) filter.

• lowpass f0[k] width[q|o|h|k]
  Second-order lowpass filter.

• highpass f0[k] width[q|o|h|k]
  Second-order highpass filter.

• bandpass_skirt f0[k] width[q|o|h|k]
  Second-order bandpass filter with constant skirt gain.

• bandpass_peak f0[k] width[q|o|h|k]
  Second-order bandpass filter with constant peak gain.

• notch f0[k] width[q|o|h|k]
  Second-order notch filter.

• allpass f0[k] width[q|o|h|k]
  Second-order allpass filter.

• eq f0[k] width[q|o|h|k] gain
  Second-order peaking filter.

• lowshelf f0[k] width[q|s|d|o|h|k] gain
  Second-order lowshelf filter.

• highshelf f0[k] width[q|s|d|o|h|k] gain
  Second-order highshelf filter.

• lowpass_transform fz[k] width_z[q] fp[k] width_p[q]
  Second-order lowpass transformation filter. Cancels the poles defined by fz and width_z and replaces them with new poles defined by fp and width_p. Gain is unity at DC.

• highpass_transform fz[k] width_z[q] fp[k] width_p[q]
  Second-order highpass transformation filter. Also known as a Linkwitz transform (see http://www.linkwitzlab.com/filters.htm#9). Same as lowpass_transform except the gain is unity at Fs/2.

• linkwitz_transform fz[k] width_z[q] fp[k] width_p[q]
  Alias for highpass_transform.

• deemph
  Compact Disc de-emphasis filter.

• biquad b0 b1 b2 a0 a1 a2
  Biquad filter.

• gain gain_dB
  Gain adjustment in decibels.

• mult multiplier
  Multiplies each sample by multiplier.

• add value
  Applies a DC shift.

• crossfeed f0[k] separation
  Simple crossfeed for headphones. Very similar to Linkwitz/Meier/CMoy/bs2b crossfeed.

• matrix4 [options] [surround_level][/surround_level_rear]
  2-to-4 channel (2 front and 2 surround) active matrix upmixer designed primarily for music. The intended speaker configuration is fronts at ±30° and surrounds between ±60° and ±120°. The front outputs replace the original input channels and the surround outputs are appended to the end of the channel list.

  For best results, the surround speakers must be calibrated to match the front speakers in both level and spectral balance. matrix4 is designed to preserve the total power of the original 2-channel input and incorrect calibration will compromise this property. After calibration, surround levels may be adjusted using the surround_level and surround_level_rear parameters. surround_level sets the level for neutral and forward steering. As steering moves from neutral to half rear, surround_level_rear gradually takes effect. This allows fine-tuning of the overall front/rear balance while still allowing full level from the surrounds when needed. The default levels are -3dB and 0dB, respectively.

  Options are given as a comma-separated list. Recognized options are:

  • show_status[=bars|text|none]
    Show a status line (slightly broken currently, but still useful for debugging).

  • matrix=v1|v2|v3
    Controls steering behavior for rear-encoded sounds. v1 does not remove anything from the front outputs. v2 adds full steering of sounds encoded to -45° in the C/S axis and restores full lateral separation of sounds encoded from 0° to -22.5° in the C/S axis in all four outputs. v3 adds full steering of sounds encoded to the left and right surround positions (L/R=±22.5° C/S=-22.5°). The default is v3.

  • shelf=gain|none[:f0[k][:pwrcmp]]
    Dynamic shelving of frequencies above f0 in surround outputs. Active when C/S is positive and gradually removed as C/S goes from 0° to -22.5°. pwrcmp is the high-frequency power compensation factor. The default values are -3dB, 500Hz, and 1, respectively.

  • lowpass=f0[k]|none
    Dynamic high-frequency rolloff (first-order lowpass shape) above f0 in surround outputs. Active when C/S is positive and gradually removed as C/S goes from 0° to -22.5°. The default is 6kHz.

  • phase_flip[=true|false]
    Dynamically flip the relative phase of the right surround output under rear steering so that stongly rear-encoded sounds are in phase. The default is true.

  • signal[=true|false]
    Toggle the effect when effect.signal() is called.

  • surround_delay=delay[s|m|S]
    Surround output delay. Generally, this should be set so that the surrounds are delayed 10-25 milliseconds relative to the fronts (measured acoustically). The default is value 15 milliseconds.

  • filter_type=filter[:stop_dB[:stop_dB]] (matrix4_mb only)
    Type of filter used for low pass sections of the filter bank. filter may be butterworth, chebyshev1, chebyshev2, or elliptic (default).

    The optional stop_dB parameter(s) set the stopband attenuation in decibels for the Chebyshev and elliptic filters. Only the first parameter is used for chebyshev1 and chebyshev2. For elliptic, the first parameter applies to the lowpass and the second to the highpass. If only one parameter is given, it applies to both stopbands. Default values are 25 for chebyshev1 and chebyshev2, and 35:50 for elliptic.

  Applying decorrelation filters to the surround outputs can further improve spatial impression over simple delay. Example:

   matrix4 surround_delay=5m -3/0
   :2,3 decorrelate -s1 -m -f0.7k -l35m 5 allpass 80 0.6 :
  

  Note that surround_delay is set to only 5ms because the decorrelation filters add approximately 10ms of delay at high frequencies. The group delay falls below 80Hz, so an additional allpass filter is added to compensate.

• matrix4_mb [options] [surround_level][/surround_level_rear]
  Like the matrix4 effect, but divides the input into 13 individually steered bands in order to improve separation of concurrent sound sources. See the matrix4 effect description for more information.

• remix selector|. ...
  Select and mix input channels into output channels. Each selector argument specifies the input channels to be mixed to produce an output channel. . selects no input channels. For example, remix 0,1 2,3 mixes input channels 0 and 1 into output channel 0, and input channels 2 and 3 into output channel 1. remix - mixes all input channels into a single output channel. The active channel selector is used as an input channel mask for the selector arguments.

• st2ms Convert stereo to mid/side.

• ms2st Convert mid/side to stereo.

• delay [-f[order]] delay[s|m|S]
  Delay line. The unit for the delay argument depends on the suffix used: s is seconds (the default), m is milliseconds, and S is samples. If delay is negative, a positive delay is applied to all channels which are not selected (except when plotting—an actual negative delay is possible in that case).

  By default, the delay is rounded to whole samples. The -f option enables fractional delay using Thiran allpass interpolation. The order argument sets the allpass filter order and may be any integer from 1 through 50. The default value is 5.

• resample [bandwidth] fs[k]
  Sinc resampler. Ignores the channel selector.

• fir [-a[offset[s|m|S]]] [input_options] [file:][~/]filter_path|coefs:list[/list...]
  Non-partitioned 64-bit direct or FFT convolution. Latency is zero for filters up to 16 taps. For longer filters, the latency is equal to the fft_len reported in verbose mode. Each list is a comma-separated list of coefficients for one filter channel. Missing values are filled with zeros.

  If -a is given and offset is zero or unspecified, channels are aligned to the filter's peak sample. Positive or negative offset values specify a fixed offset relative to the start or end of the filter, respectively.

  The input_options are useful mostly when loading raw (headerless) input files and are as follows:

  Option	Description
  -t type	Type.
  -e encoding	Encoding.
  -B/L/N	Big/little/native endian.
  -r frequency[k]	Sample rate.
  -c channels	Number of channels.

  By default, the sample rate of the filter must match that of the effect. Mismatches may be ignored by setting the sample rate to "any".

• fir_p [-a[offset[s|m|S]]] [input_options] [max_part_len] [file:][~/]filter_path|coefs:list[/list...]
  Zero-latency non-uniform partitioned 64-bit direct/FFT convolution. Usually a bit slower than the zita_convolver effect except for very long filters on some hardware. max_part_len must be a power of 2 and has a default value of 16384. Each list is a comma-separated list of coefficients for one filter channel. Missing values are filled with zeros.

  See the fir effect description for an explanation of the -a option and the input_options.

• zita_convolver [-a[offset[s|m|S]]] [input_options] [min_part_len [max_part_len]] [file:][~/]filter_path|coefs:list[/list...]
  Partitioned 32-bit FFT convolution using the zita-convolver library. Latency is equal to min_part_len (64 samples by default). {min,max}_part_len must be powers of 2 between 64 and 8192. Each list is a comma-separated list of coefficients for one filter channel. Missing values are filled with zeros.

  See the fir effect description for an explanation of the -a option and the input_options.

• hilbert [-pzc] [-a angle] taps
  Simple FIR approximation of a Hilbert transform. The number of taps must be odd. Bandwidth is controlled by the number of taps. If -p is given, the fir_p convolution engine is used instead of the default fir engine. Similarly, if -z is given, zita_convolver is used (if available). If -c is given, channels are automatically aligned to the middle tap. The -a option sets the phase shift in degrees. The default is -90°.

• decorrelate [options] [stages]
  Allpass decorrelator as described in "Frequency-Dependent Schroeder Allpass Filters" by Sebastian J. Schlecht (doi:10.3390/app10010187). With the default settings, the average group delay is around 9.5ms at high frequencies. The options are:

  Option	Description
  -m	Use the same filters for all input channels.
  -s seed	Random seed value.
  -d delay_min[s|m|S]	Minimum filter delay (default: 0.83333ms).
  -D delay_max[s|m|S]	Maximum filter delay (default: 3.12503ms).
  -f fc[k]	Damping filter center frequency (default: 1.1kHz).
  -l rt60_lf[s|m|S]	RT60 at low frequencies (default: 100ms).
  -h rt60_hf[s|m|S]	RT60 at high frequencies (default: 8ms).

  The default number of stages is 5.

• noise level[b]
  Add TPDF noise. The level argument specifies the peak level of the noise in dBFS if no suffix is given, or the effective precision in bits if the b suffix is given.

• dither [shape] [[quantize_bits] bits]
  Apply dither with optional noise shaping. The shape argument determines the type of dither and the noise shaping filter (if any):

  Name	Description
  flat	Flat TPDF with no feedback (default).
  sloped	Flat TPDF with feedback. First-order highpass response.
  sloped2	Sloped TPDF with feedback. Stronger HF emphasis than sloped.
  lipshitz	5-tap E-weighted curve from [1]. Notches around 4k and 12k.
  wan3	3-tap F-weighted curve from [2]. Notch around 4k.
  wan9	9-tap F-weighted curve from [2]. Notches around 3.5k and 12k.

  The bits argument sets the dither level in bits. The quantize_bits argument sets the number of levels to quantize to. The default setting for both is auto. If bits is not auto, dither is applied at the specified bit depth regardless of the output sample format. bits may be any number. quantize_bits must be an integer between 2 and 32. If quantize_bits is not given, it is set to the same value as bits (rounded to the nearest integer).

  Note: Currently, setting bits to auto disables dither if the effect is loaded via watch or used in ladspa_dsp.

  [1] S. P. Lipshitz, J. Vanderkooy, and R. A. Wannamaker, "Minimally Audible Noise Shaping," J. AES, vol. 39, no. 11, November 1991
  [2] R. A. Wannamaker, "Psychoacoustically Optimal Noise Shaping," J. AES, vol. 40, no. 7/8, July 1992

• ladspa_host [~/]module_path plugin_label [control ...]
  Apply a LADSPA plugin. Supports any number of input/output ports (with the exception of zero output ports). If a plugin has one or zero input ports, it will be instantiated multiple times to handle multi-channel input.

  Controls which are not explicitly set or are set to - will use default values (if available).

  The LADSPA_PATH environment variable can be used to set the search path for plugins.

• stats [ref_level]
  Display the DC offset, minimum, maximum, peak level (dBFS), RMS level (dBFS), crest factor (dB), peak count, peak sample, number of samples, and length (s) for each channel. If ref_level is given, peak and RMS levels relative to ref_level will be shown as well (dBr).

• watch [-e] [~/]path
  Load effects from a file into a sub-chain and reload if the file is modified. Other than the automatic reload, the behavior is similar to sourcing a file using the @ directive (see "Effects Files"). Some restrictions apply to automatic reload:

  • The new sub-chain must have the same output sample rate and number of channels as the previous sub-chain.
  • The new sub-chain must not require larger buffers than the previous sub-chain.

  If these conditions are not met, the new sub-chain will not be applied and an error message will be printed.

  Currently, this effect polls for file modifications once per second. Support inotify events my be added in the future. Ideally, file modifications should be atomic (i.e. by writing to a temporary file, then rename(3)-ing it over top of the original file). If this is not possible, the -e option may be given, which enforces an end-of-file marker in order to detect partially-written files. This marker, #EOF#, must be placed at the beginning of a line and may only be followed by whitespace characters.

Selector syntax

Example	Description
<empty>	all
-	all
2-	2 to n
-4	0 through 4
1,3	1 and 3
1-4,7,9-	1 through 4, 7, and 9 to n

Note: There is no difference between 1,3 and 3,1. Order is not preserved.

Filter width

The following suffixes are supported:

Suffix	Description
q	Q-factor (default).
s	Slope (shelving filters only).
d	Slope in dB/octave (shelving filters only).
o	Bandwidth in octaves.
h	Bandwidth in Hz.
k	Bandwidth in kHz.

Note: The d width suffix also changes the definition of f0 from center frequency to corner frequency (like Room EQ Wizard and the Behringer DCX2496).

Additionally, a macro is provided for constructing arbitrary-order Butterworth filters from cascaded second-order sections: bw<order>[.n], where <order> is the filter order and n is an index corresponding to a particular pair of poles. The Q-factors are always in ascending order. For example,

lowpass 1k bw6.0 lowpass 1k bw6.1 lowpass 1k bw6.2

creates a 6th-order Butterworth lowpass filter. Odd-order filters require an additional first-order section:

lowpass_1 1k lowpass 1k bw5.0 lowpass 1k bw5.1

File paths

On the command line, relative paths are relative to $PWD. Within an effects file, relative paths are relative to the directory containing said effects file. A ~/ prefix will be expanded to the contents of $HOME. The following substitutions are supported anywhere within a file path:

Sequence	Substitution
%r	Sample rate in Hz
%k	Sample rate in kHz
%c	Number of channels
%%	Literal %

Channel selectors and masks

A colon (:) followed by a selector (see "Selector syntax") specifies the input channels for effects that follow. For example,

:0,2 eq 1k 1.0 -6

will apply an eq effect to channels 0 and 2. If an effect changes the total number of channels, the last channel selector given is parsed again. Additional channels are not added unless the selector includes an unbounded range.

Channel numbers refer to the channels in the active channel mask, which is a property of the containing block. Blocks may be created using braces ({ ... }) or by sourcing a file (see "Effects files"). The channel mask is derived from the active channel selector at creation. For example,

:1,3 { :0 gain -6 :1 gain +6 }

creates a block with the mask 1,3. Within the block, :0 selects the first channel in the mask (channel 1), and :1 selects the second channel in the mask (channel 3). Channel selectors have block scope.

Channels are automatically added or removed from the active channel mask if an effect changes the total number of channels. Additional channels are always appended to the end of the channel list.

Effects files

Files may be sourced using the @ directive: @[~/]path/to/file. See "File paths" for more information about how paths are interpreted. Note that sourcing a file implicitly creates a block (see "Channel selectors and masks"). Within a file, lines in which the first non-whitespace character is # are ignored. A backslash (\) may be used to escape whitespace, #, or \. Example:

gain -4.0
# This is a comment
lowshelf 90 1s +4 eq 3k 1.5 -3

Other directives

An exclamation mark (!) allows initialization failure of the effect that follows.

FFTW wisdom

Effects utilizing FFTW3 can optionally load and save wisdom. For dsp, set $DSP_FFTW_WISDOM_PATH. ladspa_dsp uses $LADSPA_DSP_FFTW_WISDOM_PATH instead. If a path is set, FFTW plans are created with the FFTW_MEASURE flag. Accumulated wisdom is written on exit.

Examples

Read file.flac, apply a bass boost, and write to alsa device hw:2:

dsp file.flac -ot alsa -e s24_3 hw:2 lowshelf 60 0.5 +4

Plot the magnitude vs frequency response of an effects chain:

dsp -pn [effect [args]] ... | gnuplot

Implement an LR4 crossover at 2.2KHz, where output channels 0 and 1 are the left and right tweeters, and channels 2 and 3 are the left and right woofers, respectively:

dsp stereo_file.flac -ot alsa -e s32 hw:3 remix 0 1 0 1
  :0,1 highpass 2.2k 0.7071 highpass 2.2k 0.7071 :
  :2,3 lowpass 2.2k 0.7071 lowpass 2.2k 0.7071 :

Apply effects from a file:

dsp file.flac @eq.txt

LADSPA frontend

Configuration

ladspa_dsp looks for configuration files in the following directories:

• $XDG_CONFIG_HOME/ladspa_dsp
• $HOME/.config/ladspa_dsp (if $XDG_CONFIG_HOME is not set)
• /etc/ladspa_dsp

To override the default directories, set the LADSPA_DSP_CONFIG_PATH environment variable to the desired path(s) (colon-separated).

Each file that is named either config or config_<name> (where <name> is any string) is loaded as a separate plugin. The plugin label is either ladspa_dsp (for config) or ladspa_dsp:<name> (for config_<name>).

Configuration files are a simple key-value format. Leading whitespace is ignored. The valid keys are:

• input_channels
  Number of input channels. Default value is 1. May be left unset unless you want individual control over each channel.
• output_channels
  Number of output channels. Default value is 1. This parameter is not currently set automatically because the number of LADSPA ports must be known before the effects chain is built. Initialization will fail if it does not match the effects chain.
• LC_NUMERIC
  Set LC_NUMERIC to the given value while building the effects chain. Default value is C, which gives consistent number parsing behavior regardless of the system locale and LADSPA host behavior. Setting this to an empty value uses the default system locale. The special value none leaves LC_NUMERIC up to the LADSPA host (not generally recommended).
• effects_chain
  String to build the effects chain. The format is the same as an effects file, but only a single line is interpreted.

Example configuration:

# This is a comment
input_channels=1
output_channels=1
LC_NUMERIC=C
effects_chain=gain -3 lowshelf 100 1s +3 @/path/to/eq_file

Relative file paths in the effects_chain line are relative to the directory in which the configuration file resides.

The loglevel can be set to VERBOSE, NORMAL, or SILENT through the LADSPA_DSP_LOGLEVEL environment variable.

Usage example: Route alsa audio through ladspa_dsp

Put this in ~/.asoundrc:

pcm.dsp {
	type plug
	slave {
		format FLOAT
		rate unchanged
		channels unchanged
		pcm {
			type ladspa
			path "/usr/lib/ladspa"
			playback_plugins [{
				label "ladspa_dsp"
			}]
			slave.pcm {
				type plug
				slave {
					pcm "<hw_device>"
					rate unchanged
					channels unchanged
				}
			}
		}
	}
}

Replace <hw_device> with the preferred output device (hw:0, for example).

If you need individual control over each channel, you need to set the number of (output) channels:

pcm.dsp {
	type plug
	slave {
		format FLOAT
		rate unchanged
		pcm {
			type ladspa
			channels <channels>
			path "/usr/lib/ladspa"
			playback_plugins [{
				label "ladspa_dsp"
			}]
			slave.pcm {
				type plug
				slave {
					pcm "<hw_device>"
					rate unchanged
					channels unchanged
				}
			}
		}
	}
}

To make dsp the default device, append this to ~/.asoundrc:

pcm.!default {
	type copy
	slave.pcm "dsp"
}

Usage example: Route pulseaudio audio through ladspa_dsp (tested with Ubuntu 18.04; contributed by shaffenmeister)

1. Prepare .asoundrc as stated above.
2. Determine pulseaudio master sink using pacmd list sinks. Use attribute name of the pulseaudio sink you plan to use (e.g. alsa_output.pci-0000_00_14.2.analog-stereo).
3. Execute analyseplugin <path to LADSPA plugin>/ladspa_dsp.so to determine plugin name and label.
4. Run pacmd load-module module-ladspa-sink sink_name=ladspa_out sink_master=<master_sink> plugin=<plugin name> label=<plugin label>.
5. Select new LADSPA sink as system sink (Ubuntu 18.04 Desktop: Settings > Sound > Output > LADSPA_Plugin <plugin label> on <master sink>).

Example:

pacmd list sinks
analyseplugin /usr/local/lib/ladspa/ladspa_dsp.so
pacmd load-module module-ladspa-sink sink_name=ladspa_out sink_master=alsa_output.pci-0000_00_14.2.analog-stereo plugin=ladspa_dsp label=ladspa_dsp

Load LADSPA plugin as system default

To load the LADSPA module at system startup for all users include settings in /etc/pulse/default.pa:

.ifexists module-ladspa-sink.so
.nofail
load-module module-ladspa-sink sink_name=ladspa_out sink_master=<master_sink> plugin=<plugin name> label=<plugin label>
.fail
.endif

Load LADSPA plugin as user default

To load the LADSPA module at user login include settings in ~/.config/pulse/default.pa:

#!/usr/bin/pulseaudio -nF
.include /etc/pulse/default.pa
.ifexists module-ladspa-sink.so
.nofail
load-module module-ladspa-sink sink_name=ladspa_out sink_master=<master_sink> plugin=<plugin name> label=<plugin label>
.fail
.endif

Note: The resample effect cannot be used with the LADSPA frontend.

Bugs

• No support for metadata.
• Some effects do not support plotting.
• When plotting an effects chain containing the noise effect, a different random sequence is generated for each output channel regardless of whether the noise should be correlated between outputs. Summing correlated noise works correctly.

License

This software is released under the ISC license.