Contents

Chapter 1
Theory

1.1 What is ALSA?

• Advanced Linux Sound Architecture (ALSA) is a software framework and part of the Linux kernel (as a set of modules) that provides an application programming interface (API) for sound card device drivers.
• Possibilities in the Linux audio stack:

  PulseAudio / Jack / GStreamer / Xine / SDL / Phonon

  FFADO / ALSA / OSS

• Original author: Jaroslav Kysela.
• ALSA is released under the GNU General Public License (GPL) and the GNU Lesser General Public License (LGPL).
• Supported cards and driver names.

1.2 Installation

• ALSA is included in the default in most Linux kernels as a set of modules, so manual installation is not necessary.
• Udev will automatically probe your hardware at boot, loading the corresponding kernel module for your audio card.
• The loaded modules can be displayed with:
  $ lsmod | grep ’^snd’ | column -t  
  snd_hda_codec_realtek  61438   1  
  snd_hda_intel          52355   6  
  snd_hda_codec          192906  2   snd_hda_codec_realtek,snd_hda_intel  
  snd_hwdep              13602   1   snd_hda_codec  
  snd_pcm                102099  3   snd_hda_codec,snd_hda_intel  
  snd_page_alloc         18710   2   snd_pcm,snd_hda_intel  
  snd_seq_midi           13324   0  
  snd_seq_midi_event     14899   1   snd_seq_midi  
  snd_rawmidi            30144   1   snd_seq_midi  
  snd_seq                61560   2   snd_seq_midi_event,snd_seq_midi  
  snd_seq_device         14497   3   snd_seq,snd_rawmidi,snd_seq_midi  
  snd_timer              29482   2   snd_pcm,snd_seq  
  snd                    69238   21  snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec,snd_hda_intel,snd_seq_device,snd_seq_midi

1.3 Recognized audio hardware

1.4 Playback devices

1.5 PCM DAC (playback) device names

1.6 Capture devices

1.7 PCM ADC (capture) device names

1.8 Cards, devices, and subdevices

• A system can host several sound cards.
• Sound cards presents one or more devices, which are divided into one or more subdevices.
• ALSA works with logical device names, which can be hardware (real) devices or plugins.
• Hardware devices use the format hw:i,j, where i is the card number and j is the device on that card. The first device is hw:0.0.
• Plugins are refered by plughw and provides features such as sampling rate conversion, channel dupplication and sample value conversion.

1.9 (Very basic) Configuration

• If you have more than one sound card and want to use one of them by default:
  cat << EOF > ~/.asoundrc  
  # DAC  
  pcm.!default {  
      type hw  
      card Nvidia  
  }  
   
  # Mixer  
  ctl.!default {  
      type hw  
      card Nvidia  
  }  
  EOF

• Alternatively, this can be written into (with superuser priviledges):
  /etc/asound.conf

1.10 Permissions

• Users with a local login (at a virtual terminal or a display manager) have permission to play audio and change mixer levels. To allow this for a remote login, the user has to be added to the audio group.
• Membership in the audio group also allows direct access to devices. Therefore, adding a user to the audio group is not recommended, because the audio hardware can not be used for two or more users at the same time.

1.11 alsa: the deamon handler

• It should be installed by default. Location:
  $ which alsa  
  /sbin/alsa

• Allows to disable, stops, reload, etc. the ALSA driver.
  $ alsa -h  
  Usage: /sbin/alsa {unload|reload|force-unload|force-reload|suspend|resume}

1.12 amixer: a command-line mixer

• Location:
  $ which amixer  
  /usr/bin/amixer

• Mixer controls:
  $ amixer | grep control # Also works: "amixer scontrols"  
  Simple mixer control ’Master’,0               <- Master out volume control  
  Simple mixer control ’Headphone’,0            <- Stereo headphone out volume control  
  Simple mixer control ’PCM’,0                  <- Stereo DAC out volume control  
  Simple mixer control ’Front’,0                <- Stereo front (5.1) out output volume  
  Simple mixer control ’Surround’,0             <- Stereo rear out volume control  
  Simple mixer control ’Center’,0               <- Mono center out volume control  
  Simple mixer control ’LFE’,0                  <- Mono subwoofer out volume control  
  Simple mixer control ’Line’,0                 <- Stereo ADC line in volume control  
  Simple mixer control ’Line Boost’,0           <- 20dB amplified line in  
  Simple mixer control ’Mic’,0                  <- Stereo ADC microphone input line control  
  Simple mixer control ’Mic Boost’,0            <- 20dB amplified Mic  
  Simple mixer control ’IEC958’,0               <- Analog S/PDIF out volume control  
  Simple mixer control ’IEC958 Default PCM’,0   <- Digital S/PDIF out volume control  
  Simple mixer control ’Beep’,0                 <- PC "speaker" out volume control  
  Simple mixer control ’Capture’,0              <- Line in gain  
  Simple mixer control ’Capture’,1              <- S/PDIF gain  
  Simple mixer control ’Capture’,2              <- Mic gain  
  Simple mixer control ’Auto-Mute Mode’,0       <- Headphone jack inserted, speaker off  
  Simple mixer control ’Input Source’,0         <- Line | IEC958 | Mic (you can select one)  
  Simple mixer control ’Input Source’,1         <- :  
  Simple mixer control ’Input Source’,2         <- :

1.13 alsamixer: a ncursed-based mixer

1.14 speacker-test: a command-line pink noise generator

1.15 Sampling parameters

1. Sample rate: number of audio samples per second.
2. Sample format: signed, unsigned, bit/sample, etc.
3. Number of channels and data layout: refers to the number of channels and the samples interleaving.
4. Buffer size: This determines how large the hardware buffer is. It is related with the latency.

1.16 Capture limits

• Emulated:
  $ arecord --dump-hw-params > /dev/null  
  Recording WAVE ’stdin’ : Unsigned 8 bit, Rate 8000 Hz, Mono  
  HW Params of device "default": # <- Usually Pulseaudio  
  --------------------  
  ACCESS:  RW_INTERLEAVED  
  FORMAT:  U8 S16_LE S16_BE S32_LE S32_BE FLOAT_LE FLOAT_BE MU_LAW A_LAW  
  SUBFORMAT:  STD  
  SAMPLE_BITS: [8 32]  
  FRAME_BITS: [8 1024]  
  CHANNELS: [1 32]  
  RATE: [1 192000]  
  PERIOD_TIME: (5 4294967295)  
  PERIOD_SIZE: [1 1398102)  
  PERIOD_BYTES: [128 1398102)  
  PERIODS: [3 1024]  
  BUFFER_TIME: (15 4294967295]  
  BUFFER_SIZE: [3 4194304]  
  BUFFER_BYTES: [384 4194304]  
  TICK_TIME: ALL  
  --------------------  
  "Push <CTRL> + <C>"

• Real:
  $ arecord -D hw:0,0 --dump-hw-params  
  Recording WAVE ’stdin’ : Unsigned 8 bit, Rate 8000 Hz, Mono  
  HW Params of device "hw:0,0":  
  --------------------  
  ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED  
  FORMAT:  S16_LE S32_LE  
  SUBFORMAT:  STD  
  SAMPLE_BITS: [16 32]  
  FRAME_BITS: [32 64]  
  CHANNELS: 2  
  RATE: [44100 192000]  
  PERIOD_TIME: (83 185760)  
  PERIOD_SIZE: [16 8192]  
  PERIOD_BYTES: [128 65536]  
  PERIODS: [2 32]  
  BUFFER_TIME: (166 371520)  
  BUFFER_SIZE: [32 16384]  
  BUFFER_BYTES: [128 65536]  
  TICK_TIME: ALL  
  --------------------  
  arecord: set_params:1233: Sample format non available  
  Available formats:  
  - S16_LE  
  - S32_LE

1.17 Capture and playback PCM data (basic example)

Chapter 2
Practice

2.1 Exercise 1: Sample a 8-bit PCM file