Skip to content

Latest commit

 

History

History
564 lines (459 loc) · 19.3 KB

README.md

File metadata and controls

564 lines (459 loc) · 19.3 KB
Raw

About

dsp is an audio processing program with an interactive mode.

Building

Dependencies

  • GNU Make
  • pkg-config

Optional dependencies

  • fftw3: For 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.
  • libpulse-simple: For PulseAudio input/ouput support.
  • LADSPA: For the LADSPA frontend and the ladspa_host effect.
  • libltdl (libtool): For 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 ... [!] [:channel_selector]
	[@[~/]effects_file] [effect [args ...]] ...

Options

Global options

Flag Description
-h Show help text.
-b frames Set buffer size (must be given before the first input).
-R ratio Set codec maximum buffer ratio (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 instead of processing audio.
-V Enable verbose progress display.
-S Use "sequence" input combining mode.

Input/output options

Flag 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.
-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 g721_32
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 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
flac rw s16 s8 s24
caf rw s16 s8 s24 s32 float double mu-law a-law
wve rw a-law
ogg rw vorbis
mpc2k rw s16
rf64 rw s16 u8 s24 s32 float double mu-law a-law
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 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

Full 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.

  • linkwitz_transform fz[k] qz fp[k] qp
    Linkwitz transform (see http://www.linkwitzlab.com/filters.htm#9).

  • deemph
    Compact Disc de-emphasis filter.

  • biquad b0 b1 b2 a0 a1 a2
    Biquad filter.

  • gain [channel] gain
    Gain adjustment. Ignores the channel selector when the channel argument is given.

  • mult [channel] multiplier
    Multiplies each sample by multiplier. Ignores the channel selector when the channel argument is given.

  • add [channel] value
    Applies a DC shift. Ignores the channel selector when the channel argument is given.

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

  • matrix4 [options] [surround_level]
    2-to-4 channel (2 front and 2 surround) active matrix upmixer designed for plain (i.e. unencoded) stereo material. The matrix coefficients and the main ideas behind the steering behavior come from David Griesinger's published works on matrix surround.

    The intended speaker configuration is fronts at ±30° and surrounds between ±60° and ±120°. The surround speakers must be calibrated correctly in level and frequency response for best results. The surrounds should be delayed by about 10-25ms (acoustically) relative to the fronts. No frequency contouring is done internally, so applying low pass and/or shelving filters to the surround outputs is recommended:

     matrix4 surround_delay=15m -6 :2,3 lowpass_1 10k :

    The settings shown above (-6dB surround level, 15ms delay, and 10kHz rolloff) are a good starting point, but may be adjusted to taste. The default surround_level is -6dB. Applying the decorrelate effect to the surround outputs can be useful to eliminate coloration caused by comb filtering (note: adjust surround_delay to compensate for the decorrelate effect's group delay).

    The front outputs replace the original input channels and the surround outputs are appended to the end of the channel list.

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

    • no_dir_boost
      Disable directional boost of front channels.
    • show_status
      Show a status line (slightly broken currently, but still useful for debugging).
    • signal
      Toggle the effect when effect.signal() is called.
    • linear_phase (matrix4_mb only)
      Apply an FIR filter to correct the phase distortion caused by the IIR filter bank. Has no effect with matrix4. Requires the fir effect.
    • surround_delay=delay[s|m|S]
      Surround output delay. Default is zero.

  • matrix4_mb [options] [surround_level]
    Like the matrix4 effect, but divides the input into ten individually steered bands in order to improve separation of concurrent sound sources. The usage and options are the same as the matrix4 effect.

  • remix channel_selector|. ...
    Select and mix input channels into output channels. Each channel selector specifies the input channels to be mixed to produce each 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.

  • st2ms Convert stereo to mid/side.

  • ms2st Convert mid/side to stereo.

  • delay 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.

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

  • fir [file:][~/]filter_path|coefs:list[/list...]
    Non-partitioned 64-bit direct/FFT convolution. Latency is zero for filters up to 16 samples. 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.

  • fir_p [max_part_len] [file:][~/]filter_path|coefs:list[/list...]
    Zero-latency non-uniform partitioned 64-bit direct/FFT convolution. Runs slower than the zita_convolver effect, but useful if you need higher precision and/or zero latency. max_part_len must be a power of 2. Each list is a comma-separated list of coefficients for one filter channel. Missing values are filled with zeros.

  • zita_convolver [min_part_len [max_part_len]] [~/]filter_path
    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.

  • hilbert [-p] 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.

  • decorrelate [-m] [stages]
    Allpass decorrelator as described in "Frequency-Dependent Schroeder Allpass Filters" by Sebastian J. Schlecht (doi:10.3390/app10010187). If -m is given, the same filter parameters are used for all input channels. The default number of stages is 5, which results in an average group delay of about 8.5ms at high frequencies.

  • noise level
    Add TPDF noise. The level argument specifies the peak level of the noise (dBFS).

  • 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). Plugins with zero input ports will replace selected input channels with their output(s). 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).

Exclamation mark

A ! marks the effect that follows as "non-essential". If an effect is marked non-essential and it fails to initialize, it will be skipped.

Selector syntax

[[start][-[end]][,...]]
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

Width suffixes

Suffix Description
q Q-factor (default).
s Slope (shelving filters only).
d Slope in dB/octave (shelving filters only). Also changes the definition of f0 from center frequency to corner frequency (like Room EQ Wizard and the Behringer DCX2496).
o Bandwidth in octaves.
h Bandwidth in Hz.
k Bandwidth in kHz.

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.
  • The ~/ prefix will be expanded to the contents of $HOME.

Effects file syntax

  • Arguments are delimited by whitespace.
  • If the first non-whitespace character in a line is #, the line is ignored.
  • The \ character removes any special meaning of the next character.

Example:

gain -10
# This is a comment
eq 1k 1.0 +10.0 eq 3k 3.0 -4.0
lowshelf 90 0.7 +4.0

Effects files inherit a copy of the current channel selector. In other words, if an effects chain is this:

:2,4 @eq_file.txt eq 2k 1.0 -2.0

eq_file.txt will inherit the 2,4 selector, but any selector specified within eq_file.txt will not affect the eq 2k 1.0 -2.0 effect that comes after it.

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 +4.0

Plot amplitude vs frequency for a complex effects chain:

dsp -pn gain -1.5 lowshelf 60 0.7 +7.8 eq 50 2.0 -2.7 eq 100 2.0 -3.9
	eq 242 1.0 -3.8 eq 628 2.0 +2.1 eq 700 1.5 -1.0
	lowshelf 1420 0.68 -12.5 eq 2500 1.3 +3.0 eq 3000 8.0 -1.8
	eq 3500 2.5 +1.4 eq 6000 1.1 -3.4 eq 9000 1.8 -5.6
	highshelf 10000 0.7 -0.5 | gnuplot

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

dsp stereo_file.flac -ot alsa -e s32 hw:3 remix 0 0 1 1 :0,2
	lowpass 2.2k 0.707 lowpass 2.2k 0.707 :1,3 highpass 2.2k 0.707
	highpass 2.2k 0.707 :

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. Initialization will fail if this value is set incorrectly.
  • LC_NUMERIC
    Set LC_NUMERIC to the given value while building the effects chain. If the decimal separator defined by your system locale is something other than ., you should set this to C (to use . as the decimal separator) or an empty value (to use the decimal separator defined by your locale).
  • 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 -3.0 lowshelf 100 1.0s +3.0 @/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.

License

This software is released under the ISC license.