no bars cava :(

[nikaloskut]

installing cava, but bars not working...
using Arch, ALSA.
on the terminal:

my config file:

## Configuration file for CAVA.
# Remove the ; to change parameters.


[general]

# Smoothing mode. Can be 'normal', 'scientific' or 'waves'. DEPRECATED as of 0.6.0
; mode = normal

# Accepts only non-negative values.
; framerate = 60

# 'autosens' will attempt to decrease sensitivity if the bars peak. 1 = on, 0 = off
# new as of 0.6.0 autosens of low values (dynamic range)
# 'overshoot' allows bars to overshoot (in % of terminal height) without initiating autosens. DEPRECATED as of 0.6.0
; autosens = 1
; overshoot = 20

# Manual sensitivity in %. If autosens is enabled, this will only be the initial value.
# 200 means double height. Accepts only non-negative values.
; sensitivity = 100

# The number of bars (0-512). 0 sets it to auto (fill up console).
# Bars' width and space between bars in number of characters.
; bars = 0
; bar_width = 2
; bar_spacing = 1
# bar_height is only used for output in "noritake" format
; bar_height = 32

# For SDL width and space between bars is in pixels, defaults are:
; bar_width = 20
; bar_spacing = 5

# sdl_glsl have these default values, they are only used to calculate max number of bars.
; bar_width = 1
; bar_spacing = 0


# Lower and higher cutoff frequencies for lowest and highest bars
# the bandwidth of the visualizer.
# Note: there is a minimum total bandwidth of 43Mhz x number of bars.
# Cava will automatically increase the higher cutoff if a too low band is specified.
; lower_cutoff_freq = 50
; higher_cutoff_freq = 10000


# Seconds with no input before cava goes to sleep mode. Cava will not perform FFT or drawing and
# only check for input once per second. Cava will wake up once input is detected. 0 = disable.
; sleep_timer = 0


[input]

# Audio capturing method. Possible methods are: 'fifo', 'portaudio', 'pipewire', 'alsa', 'pulse', 'sndio', 'oss', 'jack' or 'shmem'
# Defaults to 'oss', 'pipewire', 'sndio', 'jack', 'pulse', 'alsa', 'portaudio' or 'fifo', in that order, dependent on what support cava was built with.
# On Mac it defaults to 'portaudio' or 'fifo'
# On windows this is automatic and no input settings are needed.
#
# All input methods uses the same config variable 'source'
# to define where it should get the audio.
#
# For pulseaudio and pipewire 'source' will be the source. Default: 'auto', which uses the monitor source of the default sink
# (all pulseaudio sinks(outputs) have 'monitor' sources(inputs) associated with them).
#
# For pipewire 'source' will be the object name or object.serial of the device to capture from.
# Both input and output devices are supported.
#
# For alsa 'source' will be the capture device.
# For fifo 'source' will be the path to fifo-file.
# For shmem 'source' will be /squeezelite-AA:BB:CC:DD:EE:FF where 'AA:BB:CC:DD:EE:FF' will be squeezelite's MAC address
#
# For sndio 'source' will be a raw recording audio descriptor or a monitoring sub-device, e.g. 'rsnd/2' or 'snd/1'. Default: 'default'.
# README.md contains further information on how to setup CAVA for sndio.
#
# For oss 'source' will be the path to a audio device, e.g. '/dev/dsp2'. Default: '/dev/dsp', i.e. the default audio device.
# README.md contains further information on how to setup CAVA for OSS on FreeBSD.
#
# For jack 'source' will be the name of the JACK server to connect to, e.g. 'foobar'. Default: 'default'.
# README.md contains further information on how to setup CAVA for JACK.
#
; method = pulse
; source = auto

; method = pipewire
; source = auto

; method = alsa
; source = hw:Loopback,1

; method = fifo
; source = /tmp/mpd.fifo

; method = shmem
; source = /squeezelite-AA:BB:CC:DD:EE:FF

; method = portaudio
; source = auto

; method = sndio
; source = default

; method = oss
; source = /dev/dsp

; method = jack
; source = default
# The options 'sample_rate', 'sample_bits', 'channels' and 'autoconnect' can be configured for some input methods:
#   sample_rate: fifo, pipewire, sndio, oss
#   sample_bits: fifo, pipewire, sndio, oss
#   channels:    sndio, oss, jack
#   autoconnect: jack
# Other methods ignore these settings.
#
# For 'sndio' and 'oss' they are only preferred values, i.e. if the values are not supported
# by the chosen audio device, the device will use other supported values instead.
# Example: 48000, 32 and 2, but the device only supports 44100, 16 and 1, then it
# will use 44100, 16 and 1.
#
; sample_rate = 44100
; sample_bits = 16
; channels = 2
; autoconnect = 2


[output]

# Output method. Can be 'ncurses', 'noncurses', 'raw', 'noritake', 'sdl'
# or 'sdl_glsl'.
# 'noncurses' (default) uses a buffer and cursor movements to only print
# changes from frame to frame in the terminal. Uses less resources and is less
# prone to tearing (vsync issues) than 'ncurses'.
#
# 'raw' is an 8 or 16 bit (configurable via the 'bit_format' option) data
# stream of the bar heights that can be used to send to other applications.
# 'raw' defaults to 200 bars, which can be adjusted in the 'bars' option above.
#
# 'noritake' outputs a bitmap in the format expected by a Noritake VFD display
#  in graphic mode. It only support the 3000 series graphical VFDs for now.
#
# 'sdl' uses the Simple DirectMedia Layer to render in a graphical context.
# 'sdl_glsl' uses SDL to create an OpenGL context. Write your own shaders or
# use one of the predefined ones.
; method = noncurses

# Orientation of the visualization. Can be 'bottom', 'top', 'left', 'right' or
# 'horizontal'. Default is 'bottom'. 'left and 'right' are only supported on sdl
# and ncruses output. 'horizontal' (bars go up and down from center) is only supported
# on noncurses output.
# Note: many fonts have weird or missing glyphs for characters used in orientations
# other than 'bottom', which can make output not look right.
; orientation = bottom

# Visual channels. Can be 'stereo' or 'mono'.
# 'stereo' mirrors both channels with low frequencies in center.
# 'mono' outputs left to right lowest to highest frequencies.
# 'mono_option' set mono to either take input from 'left', 'right' or 'average'.
# set 'reverse' to 1 to display frequencies the other way around.
; channels = stereo
; mono_option = average
; reverse = 0

# Raw output target. A fifo will be created if target does not exist.
; raw_target = /dev/stdout

# Raw data format. Can be 'binary' or 'ascii'.
; data_format = binary

# Binary bit format, can be '8bit' (0-255) or '16bit' (0-65530).
; bit_format = 16bit

# Ascii max value. In 'ascii' mode range will run from 0 to value specified here
; ascii_max_range = 1000

# Ascii delimiters. In ascii format each bar and frame is separated by a delimiters.
# Use decimal value in ascii table (i.e. 59 = ';' and 10 = '\n' (line feed)).
; bar_delimiter = 59
; frame_delimiter = 10

# sdl window size and position. -1,-1 is centered.
; sdl_width = 1000
; sdl_height = 500
; sdl_x = -1
; sdl_y= -1
; sdl_full_screen = 0

# set label on bars on the x-axis. Can be 'frequency' or 'none'. Default: 'none'
# 'frequency' displays the lower cut off frequency of the bar above.
# Only supported on ncurses and noncurses output.
; xaxis = none
 
# enable synchronized sync. 1 = on, 0 = off
# removes flickering in alacritty terminal emulator.
# defaults to off since the behaviour in other terminal emulators is unknown
; synchronized_sync = 0

# Shaders for sdl_glsl, located in $HOME/.config/cava/shaders
; vertex_shader = pass_through.vert
; fragment_shader = bar_spectrum.frag

; for glsl output mode, keep rendering even if no audio
; continuous_rendering = 0

# disable console blank (screen saver) in tty
# (Not supported on FreeBSD)
; disable_blanking = 0

# show a flat bar at the bottom of the screen when idle, 1 = on, 0 = off
; show_idle_bar_heads = 1

# show waveform instead of frequency spectrum, 1 = on, 0 = off
; waveform = 0

[color]
# Colors can be one of seven predefined: black, blue, cyan, green, magenta, red, white, yellow.
# Or defined by hex code '#xxxxxx' (hex code must be within ''). User defined colors requires
# a terminal that can change color definitions such as Gnome-terminal or rxvt.
# default is to keep current terminal color
; background = default
; foreground = default

# SDL and sdl_glsl only support hex code colors, these are the default:
; background = '#111111'
; foreground = '#33ffff'


# Gradient mode, only hex defined colors are supported,
# background must also be defined in hex or remain commented out. 1 = on, 0 = off.
# You can define as many as 8 different colors. They range from bottom to top of screen
; gradient = 0
; gradient_color_1 = '#59cc33'
; gradient_color_2 = '#80cc33'
; gradient_color_3 = '#a6cc33'
; gradient_color_4 = '#cccc33'
; gradient_color_5 = '#cca633'
; gradient_color_6 = '#cc8033'
; gradient_color_7 = '#cc5933'
; gradient_color_8 = '#cc3333'



[smoothing]

# Percentage value for integral smoothing. Takes values from 0 - 100.
# Higher values means smoother, but less precise. 0 to disable.
# DEPRECATED as of 0.8.0, use noise_reduction instead
; integral = 77

# Disables or enables the so-called "Monstercat smoothing" with or without "waves". Set to 0 to disable.
; monstercat = 0
; waves = 0

# Set gravity percentage for "drop off". Higher values means bars will drop faster.
# Accepts only non-negative values. 50 means half gravity, 200 means double. Set to 0 to disable "drop off".
# DEPRECATED as of 0.8.0, use noise_reduction instead
; gravity = 100


# In bar height, bars that would have been lower that this will not be drawn.
# DEPRECATED as of 0.8.0
; ignore = 0

# Noise reduction, int 0 - 100. default 77
# the raw visualization is very noisy, this factor adjusts the integral and gravity filters to keep the signal smooth
# 100 will be very slow and smooth, 0 will be fast but noisy.
; noise_reduction = 77


[eq]

# This one is tricky. You can have as much keys as you want.
# Remember to uncomment more than one key! More keys = more precision.
# Look at readme.md on github for further explanations and examples.
; 1 = 1 # bass
; 2 = 1
; 3 = 1 # midtone
; 4 = 1
; 5 = 1 # treble

[karlstav]

config file is just default is it not?

can you describe your audio setup? pulseaudio, pipewire, bluetooth, usb?

[nikaloskut]

config is default, using pulseaudio

[karlstav]

can you give me the output of pacmd list-sinks and pacmd list-sources

[nikaloskut]

meow@Arch ~> pacmd list-sinks
1 sink(s) available.
  * index: 0
	name: <alsa_output.pci-0000_00_14.2.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 9039
	volume: front-left: 19662 /  30% / -31.37 dB,   front-right: 19662 /  30% / -31.37 dB
	       balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max request: 0 KiB
	max rewind: 0 KiB
	monitor source: 0
	sample spec: s16le 2-канальный 4410
	channel map: front-left,front-right
	            Стерео
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
	card: 1 <alsa_card.pci-0000_00_14.2>
	module: 7
	properties:
		alsa.resolution_bits = "16"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC233 Analog"
		alsa.id = "ALC233 Analog"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "1"
		alsa.card_name = "HD-Audio Generic"
		alsa.long_card_name = "HD-Audio Generic at 0xf0d64000 irq 16"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:14.2"
		sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card1"
		device.bus = "pci"
		device.vendor.id = "1022"
		device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
		device.product.id = "780d"
		device.product.name = "FCH Azalia Controller"
		device.form_factor = "internal"
		device.string = "front:1"
		device.buffering.buffer_size = "352768"
		device.buffering.fragment_size = "176384"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Аналоговый стерео"
		device.description = "Встроенное аудио Аналоговый стерео"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-output-speaker: Динамики (priority 10000, latency offset 0 usec, available: unknown)
			properties:
				device.icon_name = "audio-speakers"
		analog-output-headphones: Наушники (priority 9900, latency offset 0 usec, available: no)
			properties:
				device.icon_name = "audio-headphones"
	active port: <analog-output-speaker>




meow@Arch ~> pacmd list-sources
2 source(s) available.
    index: 0
	name: <alsa_output.pci-0000_00_14.2.analog-stereo.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 1030
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	       balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s16le 2-канальный 4410
	channel map: front-left,front-right
	            Стерео
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
	monitor_of: 0
	card: 1 <alsa_card.pci-0000_00_14.2>
	module: 7
	properties:
		device.description = "Monitor of Встроенное аудио Аналоговый стерео"
		device.class = "monitor"
		alsa.card = "1"
		alsa.card_name = "HD-Audio Generic"
		alsa.long_card_name = "HD-Audio Generic at 0xf0d64000 irq 16"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:14.2"
		sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card1"
		device.bus = "pci"
		device.vendor.id = "1022"
		device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
		device.product.id = "780d"
		device.product.name = "FCH Azalia Controller"
		device.form_factor = "internal"
		device.string = "1"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
  * index: 1
	name: <alsa_input.pci-0000_00_14.2.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 9039
	volume: front-left: 10387 /  16% / -48.00 dB,   front-right: 10387 /  16% / -48.00 dB
	       balance 0.00
	base volume: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s16le 2-канальный 4410
	channel map: front-left,front-right
	            Стерео
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
	card: 1 <alsa_card.pci-0000_00_14.2>
	module: 7
	properties:
		alsa.resolution_bits = "16"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC233 Analog"
		alsa.id = "ALC233 Analog"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "1"
		alsa.card_name = "HD-Audio Generic"
		alsa.long_card_name = "HD-Audio Generic at 0xf0d64000 irq 16"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:14.2"
		sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card1"
		device.bus = "pci"
		device.vendor.id = "1022"
		device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
		device.product.id = "780d"
		device.product.name = "FCH Azalia Controller"
		device.form_factor = "internal"
		device.string = "front:1"
		device.buffering.buffer_size = "352768"
		device.buffering.fragment_size = "176384"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Аналоговый стерео"
		device.description = "Встроенное аудио Аналоговый стерео"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-input-internal-mic: Встроенный микрофон (priority 8900, latency offset 0 usec, available: unknown)
			properties:
				device.icon_name = "audio-input-microphone"
		analog-input-mic: Микрофон (priority 8700, latency offset 0 usec, available: no)
			properties:
				device.icon_name = "audio-input-microphone"
	active port: <analog-input-internal-mic>
meow@Arch ~> 

[karlstav]

this looks like very much straight forward. you could try to force pulse input, remove the ; in front of method = pulse in the config

[nikaloskut]

Thank you very much! Everything worked! <3

[karlstav]

nice, this issue here is that you most likely have the pipewire dev ifles installed so cava was built with pipewire support as well as pulse. Pipewire gets priority before pulse, even though you are not using pipewire. I should probably try to find a better solution as I would guess more people would stumble upon the same issue