ALSA without PulseAudio and PipeWire

igorzwx · 2024-05-30 03:06:21

It is not difficult to remove PulseAudio and PipeWire.
The problem is how to configure a software mixer for ALSA.

Such mixer might be needed, if one wants to use ALSA just like MacOS users are using the sound system on MacBooks: for video conferencing with Discord or similar app, or playing HiRes audio files without distortions produced by crappy resamplers. This implies that the desired software mixer should have a high quality real-time codec (resampler and format converter).

These problems can be solved through the help of the fftrate plugin
https://github.com/PetrovSE/fftrate

Debian packages:

libasound2-plugin-fftrate_1.6.3_amd64.deb
alsa-config-utils_1.6.3_amd64.deb
pcm-utils_1.6.3_amd64.deb

They provide:

libasound_module_rate_fftrate.so — ALSA plugin (FFT resampler/converter)
arateconf  —  utility to generate ~/.asoundrc  
pcm_conv  —  standalone FFT resampler/converter

How to compile fftrate plugin

Install "build-essential", etc.

sudo apt update
sudo apt-get install build-essential git debhelper
sudo apt install libasound2-dev libasound2-plugins swh-plugins

Install gcc-10 and g++-10 from Devuan's oldstable repositories.

$ cat /etc/apt/sources.list | grep oldstable
# oldstable repositories
deb http://deb.devuan.org/merged oldstable main contrib non-free
deb-src http://deb.devuan.org/merged oldstable main contrib non-free

Enable the oldstable repositories and update

sudo apt update

$ apt search g* | grep "gcc-10/\|g++-10/"
g++-10/oldstable,now 10.2.1-6 amd64 
gcc-10/oldstable,now 10.2.1-6 amd64

sudo apt install gcc-10 g++-10

$ whereis gcc-10 g++-10
gcc-10: /usr/bin/gcc-10 
g++-10: /usr/bin/g++-10

To compile the code you should configure "make" to use gcc-10 and g++-10, instead of the default gcc-12 and g++-12.
To build the deb packages, you have to specify "compatibility level 13" in debian/compat

$ man debhelper-compat-upgrade-checklist | grep "recommended mode"
          v13 This is the recommended mode of operation.

Compilation

mkdir petrov-fftrate
cd petrov-fftrate
git clone https://github.com/PetrovSE/fftrate.git

Now you have to edit two files inside the package "fftrate":

./fftrate/src/lib/makedef.mk
./fftrate/packets/debian/compat

Enable gcc-10 and g++-10:

$ cat ./fftrate/src/lib/makedef.mk | grep gcc -A1
CC      = gcc-10
CPP     = g++-10

Change compat to "13":

$ cat ./fftrate/packets/debian/compat
13

Now you can compile:

cd  ./fftrate/packets

$ ls -1
Makefile
debian
etc
mk_dpkg

Compilte the code:

make

Make debian packages:

./mk_dpkg

$ ls ../ | grep .deb
alsa-config-utils_1.6.3_amd64.deb
libasound2-plugin-fftrate_1.6.3_amd64.deb
pcm-utils_1.6.3_amd64.deb

How to install fftrate

Install dependencies (if they are not already installed):

sudo apt install libasound2 libasound2-plugins swh-plugins

Install fftrate:

sudo dpkg -i libasound2-plugin-fftrate_1.6.3_amd64.deb alsa-config-utils_1.6.3_amd64.deb pcm-utils_1.6.3_amd64.deb

Open /etc/fftrate.conf with a text editor:

sudo nano /etc/fftrate.conf

Enable fft resampler:

$ cat /etc/fftrate.conf | grep "Transform type" -A3
# Transform type
# Available: dct, fft (default: dct)
#transform = dct
transform = fft

Run arateconf (interactive mode) to generate ~/.asoundrc

arateconf

Press 0 to select the first sound card.
Press 1 to select the second sound card.
Press F to change format

For Intel HDA codec on motherboard, try 32bit 192000Hz
For USB headsets, try 16bit 48Hz

Run TEST (press T).
If all tests have been passed successfully, save the setting (press S) and exit (ESC).

EXAMPLE: Intel HDA

===============
|  Main menu  |
---------------

    Curr. | Used | Play (def) | Rec (def) | Available cards
----------|------|------------|-----------|-----------------------------------
0 - >>>>> |  *   |  *     *   |  *    *   | HDA-Intel - HDA Intel PCH
1 -       |  *   |  *         |  *        | USB-Audio - iMic USB audio system

U - Toggle used flag
P - Set this device as default player
R - Set this device as default recorder

O - Output device: PCH,0
I - Input device : PCH,0

F - Format: 192000 Hz, 2 ch, 'S32_LE'
C - Converter: fftrate

A - Show all plugins [ ]
M - Plug-ins:
 [X] Convert,  [ ] Expand, [X] Asym
 [ ] Play Vol, [X] Dmix
 [ ] Rec. Vol, [X] Dsnoop
 [ ] Phonon,   [ ] Normalizator

T - Test
S - Save to '/home/devuan/.asoundrc'
X - Delete '/home/devuan/.asoundrc'

ESC - Exit

EXAMPLE: iMic USB

===============
|  Main menu  |
---------------

    Curr. | Used | Play (def) | Rec (def) | Available cards
----------|------|------------|-----------|-----------------------------------
0 -       |  *   |  *     *   |  *    *   | HDA-Intel - HDA Intel PCH
1 - >>>>> |  *   |  *         |  *        | USB-Audio - iMic USB audio system

U - Toggle used flag
P - Set this device as default player
R - Set this device as default recorder

O - Output device: system,0
I - Input device : system,0

F - Format: 48000 Hz, 2 ch, 'S16_LE'
C - Converter: fftrate

A - Show all plugins [ ]
M - Plug-ins:
 [X] Convert,  [ ] Expand, [X] Asym
 [ ] Play Vol, [X] Dmix
 [ ] Rec. Vol, [X] Dsnoop
 [ ] Phonon,   [ ] Normalizator

T - Test
S - Save to '/home/devuan/.asoundrc'
X - Delete '/home/devuan/.asoundrc'

ESC - Exit

TEST

> t
Testing H/W compatibility ...

Device: PCH
Output:
 Open device "hw:PCH,0" ... Ok.
 Set rate 192000 Hz ... Ok.
 Set channels 2 ... Ok.
 Set format 'S32_LE' ... Ok.
 Set buffer size 30720 -> 30720
 Set period size 7680 -> 7680
Input:
 Open device "hw:PCH,0" ... Ok.
 Set rate 192000 Hz ... Ok.
 Set format 'S32_LE' ... Ok.
 Set buffer size 30720 -> 30720
 Set period size 7680 -> 7680

Device: system
Output:
 Open device "hw:system,0" ... Ok.
 Set rate 48000 Hz ... Ok.
 Set channels 2 ... Ok.
 Set format 'S16_LE' ... Ok.
 Set buffer size 7680 -> 7680
 Set period size 1920 -> 1920
Input:
 Open device "hw:system,0" ... Ok.
 Set rate 48000 Hz ... Ok.
 Set format 'S16_LE' ... Ok.
 Set buffer size 7680 -> 7680
 Set period size 1920 -> 1920

Press any key ...

S - Save to '/home/devuan/.asoundrc'

> s
Saving config file...
Test: 'default' ... Ok.
Test: 'default' ... Ok.
Ok.

Testing playback/recording

$ aplay *.wav
Playing WAVE 'audio_test_48kHz_16bit.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Output: 192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Rates:  48000 --> 192000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ arecord -V stereo -f S16_LE -r 48000 -c 2 1-fft-exp_rec_alsa.wav
Recording WAVE '1-fft-exp_rec_alsa.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  192000 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ aplay 1-fft-exp_rec_alsa.wav
Playing WAVE '1-fft-exp_rec_alsa.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Output: 192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Rates:  48000 --> 192000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

Now you can install apulse and run discord

apulse discord

You may also need ALSA mixer

alsamixer

man alsamixer

Last edited by igorzwx (2024-05-31 00:40:14)

swanson · 2024-05-30 08:58:22

Thank you! Looks very interesting.

igorzwx · 2025-02-15 22:57:34

EDITED

BEFORE running "arateconf", you should close all audio applications.
The "TEST" function of "arateconf" will fail, if your "audio device" is already in use.
When pulseaudio and pipewire are removed, you can easily free your "audio devices" with "fuser".
For example:

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:   igor       8248 F...m firefox-esr
/dev/snd/seq:
/dev/snd/timer:      igor       8248 f.... firefox-esr

"mate-settings-daemon" and "mate-volume-control-status-icon" need not to be killed.
But "firefox-esr" should be killed to free the "audio device":

$ fuser -ikv $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
Kill process 2261 ? (y/N) n
Kill process 2310 ? (y/N) n
/dev/snd/pcmC0D0p:   igor       8248 F...m firefox-esr
Kill process 8248 ? (y/N) y
/dev/snd/timer:      igor       8248 f.... firefox-esr
Kill process 8248 ? (y/N) y
Could not kill process 8248: No such process

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:
/dev/snd/seq:
/dev/snd/timer:

Now, when audio devices are free, you can run "arateconf".

kapqa wrote:

if i press "T", there is "failed".
Input:
 Open device "hw:C1,0" ... Failed!

It failed to open "Input", because you configured your "Cambridge USB DAC" for recording.
"DAC" means "digital-to-analog converter",
see _https://en.wikipedia.org/wiki/Digital-to-analog_converter.
As the name suggests, it is not supposed to be used to record "analog sound".

For "Output", you also got an error message:

Set format 'S32_LE' ... Unsupported!

It means that your Cambridge DAC does not support 'S32_LE' (it is now default for playback),
or ALSA driver (snd_usb_audio) for your USB DAC does not support 32bit.
Try 'S16_LE', or else.
Read the manual again and change 'S32_LE' to 'S16_LE'.
Press "F" to change "format".

======================
|  Sound parameters  |
----------------------
S - Sample rate = 48000 Hz
C - Channels    = 2
F - Format: 'S32_LE'

A - Alignment buffer and period [X]
B - Set buffer multiplier = 1
M - Set period multiplier = 1

P - Play ampl. = 0 dB
R - Rec. ampl. = 0 dB

ESC - Return to main menu
> f

==============================
|  Available sample formats  |
------------------------------
0 - U8
1 - S16_LE
2 - S24_3LE
3 - S24_LE
4 - S32_LE
5 - FLOAT_LE
6 - FLOAT64_LE

ESC - Return to main menu
>

Your Intel HDA codec (Generic_1) does support 48kHz 32bit.
It should also support 192kHz 32bit.
This may provide much better sound quality than your "Cambridge DAC" (it is likely to be 48kHz 16bit and nothing more with ALSA).
You can test different "sample rates" and "formats" with "arateconf".

You can also check your "proc" (cards, hw_params), for example:

$ cat /proc/asound/cards
 0 [system         ]: USB-Audio - iMic USB audio system
                      Griffin Technology, Inc iMic USB audio system at usb-0000:00:1a.0-1.3.4, full s

$ cat /proc/asound/card*/pcm*p/sub*/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1920
buffer_size: 7680

$ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1920
buffer_size: 7680

NOTE: "arateconf" is simply a tool to configure ALSA,
that is, a tool to generate an ALSA config file "~/.asoundrc".
It takes data from ALSA and from Linux (proc, etc.).

Notable Realtek products include ... audio codecs (AC'97 and Intel HD Audio ).
_https://en.wikipedia.org/wiki/Realtek#Notable_products

ALSA detected an Intel HDA codec on your motherboard. It has a "generic driver" for it.
It should work without problems.
You may select Intel HDA codec as default for recording.
If you connect new USB audio devices, you have to create a new ALSA config with "arateconf".

Last edited by igorzwx (2025-02-16 21:38:24)

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#1 2024-05-30 03:06:21

ALSA without PulseAudio and PipeWire

#2 2024-05-30 08:58:22

Re: ALSA without PulseAudio and PipeWire

#3 2025-02-15 22:57:34

Re: ALSA without PulseAudio and PipeWire

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