owntone-server/README_ALSA.md
2020-07-15 13:41:00 +01:00

20 KiB

forked-daapd and ALSA

ALSA is one of the main output configuration options for local audio; when using ALSA you will typically let the system select the soundcard on your machine as the default device/sound card - a mixer associated with the ALSA device is used for volume control. However if your machine has multiple sound cards and your system chooses the wrong playback device, you will need to manually select the card and mixer to complete the forked daapd configuration.

Quick introduction to ALSA devices

ALSA devices can be addressed in a number ways but traditionally we have referred to them using the hardware prefix, card number and optionally device number with something like hw:0 or hw:0,1. In ALSA configuration terms card X, device Y is known as hw:X,Y.

ALSA has other prefixes for each card and most importantly plughw. The plughw performs transparent sample format and sample rate conversions and maybe a better choice for many users rather than hw: which would fail when provided unsupported audio formats/sample rates.

Alternative ALSA names can be used to refer to physical ALSA devices and can be useful in a number of ways:

  • more descriptive rather than being a card number
  • consistent for USB numeration - USB ALSA devices may not have the same card number across reboots/reconnects

The ALSA device information required for configuration the server can be deterined using aplay, as described in the rest of this document, but forked-daapd can also assist; when configured to log at INFO level the following information is provided during startup:

laudio: Available ALSA playback mixer(s) on hw:0 CARD=Intel (HDA Intel): 'Master' 'Headphone' 'Speaker' 'PCM' 'Mic' 'Beep'
laudio: Available ALSA playback mixer(s) on hw:1 CARD=E30 (E30): 'E30 '
laudio: Available ALSA playback mixer(s) on hw:2 CARD=Seri (Plantronics Blackwire 3210 Seri): 'Sidetone' 'Headset'

The CARD= string is the alternate ALSA name for the device and can be used in place of the traditional hw:x name.

On this machine the server reports that it can see the onboard HDA Intel sound card and two additional sound cards: a Topping E30 DAC and a Plantronics Headset which are both USB devices. We can address the first ALSA device as hw:0 or hw:CARD=Intel or hw:Intel or plughw:Intel, the second ALSA device as hw:1 or hw:E30 and so forth. The latter 2 devices being on USB will mean that hw:1 may not always refer to hw:E30 and thus in such a case using the alternate name is useful.

Configuring the server

forked-daapd can support a single ALSA device or multiple ALSA devices.

# example audio section for server for a single soundcard
audio {
    nickname = "Computer"
    type = "alsa"

    card = "hw:1"           # defaults to 'default'
    mixer = "Analogue"      # defaults to 'PCM' or 'Master'
    mixer_device = "hw:1"   # defaults to same as 'card' value
}

Multiple devices can be made available to forked-daapd using seperate alsa { .. } sections.

audio {
    type = "alsa"
}

alsa "hw:1" {
    nickname = "Computer"
    mixer = "Analogue"
    mixer_device = "hw:1"
}

alsa "hw:2" {
    nickname = "Second ALSA device"
}

NB: When introducing alsa { .. } section(s) the ALSA specific configuration in the audio { .. } section will be ignored.

If there is only one sound card, verify if the default sound device is correct for playback, we will use the aplay utility.

# generate some audio if you don't have a wav file to hand
$ sox -n -c 2 -r 44100 -b 16 -C 128 /tmp/sine441.wav synth 30 sin 500-100 fade h 0.2 30 0.2

$ aplay -Ddefault /tmp/sine441.wav

If you can hear music played then you are good to use default for the server configuration. If you can not hear anything from the aplay firstly verify (using alsamixer) that the sound card is not muted. If the card is not muted AND there is no sound you can try the options below to determine the card and mixer for configuring the server.

Automatically Determine ALL relevant the sound card information

As shown above, forked-daapd can help, consider the information that logged:

laudio: Available ALSA playback mixer(s) on hw:0 CARD=Intel (HDA Intel): 'Master' 'Headphone' 'Speaker' 'PCM' 'Mic' 'Beep'
laudio: Available ALSA playback mixer(s) on hw:1 CARD=E30 (E30): 'E30 '
laudio: Available ALSA playback mixer(s) on hw:2 CARD=Seri (Plantronics Blackwire 3210 Seri): 'Sidetone' 'Headset'

Using the information above, we can see 3 soundcards that we could use with forked-daap with the first soundcard having a number of seperate mixer devices (volume control) for headphone and the interal speakers - we'll configure the server to use both these and also the E30 device. The server configuration for theese multiple outputs would be:

# using ALSA device alias where possible

alsa "hw:Intel" {
    nickname = "Computer - Speaker"
    mixer = "Speaker"
}

alsa "hw:Intel" {
    nickname = "Computer - Headphones"
    mixer = "Headphone"
}

alsa "plughw:E30" {
    # this E30 device only support S32_LE so we can use the 'plughw' prefix to
    # add transparent conversion support of more common S16/S24_LE formats

    nickname = "E30 DAC"
    mixer = "E30 "
    mixer_device = "hw:E30"
}

NB: it is troublesome to use hw or plughw ALSA addressing when running forked-daapd on a machine with pulseaudio and if you wish to use refer to ALSA devices directly that you stop pulseaudio.

Manually Determining the sound cards you have / ALSA can see

The example below is how I determined the correct sound card and mixer values for a Raspberry Pi that has an additional DAC card (hat) mounted. Of course using the log output from the server would have given the same results.

Use aplay -l to list all the sound cards and their order as known to the system - you can have multiple card X, device Y entries; some cards can also have multiple playback devices such as the RPI's onboard soundcard which feeds both headphone (card 0, device 0) and HDMI (card 0, device 1).

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA]
  Subdevices: 6/7
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
  Subdevice #2: subdevice #2
  Subdevice #3: subdevice #3
  Subdevice #4: subdevice #4
  Subdevice #5: subdevice #5
  Subdevice #6: subdevice #6
card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 ALSA [bcm2835 IEC958/HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: IQaudIODAC [IQaudIODAC], device 0: IQaudIO DAC HiFi pcm512x-hifi-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

On this machine we see the second sound card installed, an IQaudIODAC dac hat, and identified as card 1 device 0. This is the playback device we want to be used by the server.

hw:1,0 is the IQaudIODAC that we want to use - we verify audiable playback through that sound card using aplay -Dhw:1 /tmp/sine441.wav. If the card has only one device, we can simply refer to the sound card using hw:X so in this case where the IQaudIODAC only has one device, we can refer to this card as hw:1 or hw:1,0.

Use aplay -L to get more information about the PCM devices defined on the system.

$ aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
default:CARD=ALSA
    bcm2835 ALSA, bcm2835 ALSA
    Default Audio Device
sysdefault:CARD=ALSA
    bcm2835 ALSA, bcm2835 ALSA
    Default Audio Device
dmix:CARD=ALSA,DEV=0
    bcm2835 ALSA, bcm2835 ALSA
    Direct sample mixing device
dmix:CARD=ALSA,DEV=1
    bcm2835 ALSA, bcm2835 IEC958/HDMI
    Direct sample mixing device
dsnoop:CARD=ALSA,DEV=0
    bcm2835 ALSA, bcm2835 ALSA
    Direct sample snooping device
dsnoop:CARD=ALSA,DEV=1
    bcm2835 ALSA, bcm2835 IEC958/HDMI
    Direct sample snooping device
hw:CARD=ALSA,DEV=0
    bcm2835 ALSA, bcm2835 ALSA
    Direct hardware device without any conversions
hw:CARD=ALSA,DEV=1
    bcm2835 ALSA, bcm2835 IEC958/HDMI
    Direct hardware device without any conversions
plughw:CARD=ALSA,DEV=0
    bcm2835 ALSA, bcm2835 ALSA
    Hardware device with all software conversions
plughw:CARD=ALSA,DEV=1
    bcm2835 ALSA, bcm2835 IEC958/HDMI
    Hardware device with all software conversions
default:CARD=IQaudIODAC
    IQaudIODAC, 
    Default Audio Device
sysdefault:CARD=IQaudIODAC
    IQaudIODAC, 
    Default Audio Device
dmix:CARD=IQaudIODAC,DEV=0
    IQaudIODAC, 
    Direct sample mixing device
dsnoop:CARD=IQaudIODAC,DEV=0
    IQaudIODAC, 
    Direct sample snooping device
hw:CARD=IQaudIODAC,DEV=0
    IQaudIODAC, 
    Direct hardware device without any conversions
plughw:CARD=IQaudIODAC,DEV=0
    IQaudIODAC, 
    Hardware device with all software conversions

For the server configuration, we will use:

audio {
    nickname = "Computer"
    type = "alsa"
    card="hw:1"
    # mixer=TBD
    # mixer_device=TBD
}

Mixer name

Once you have the card number (determined from aplay -l) we can inspect/confirm the name of the mixer that can be used for playback (it may NOT be PCM as expected by the server). In this example, the card 1 is of interest and thus we use -c 1 with the following command:

$ amixer -c 1 
Simple mixer control 'DSP Program',0
  Capabilities: enum
  Items: 'FIR interpolation with de-emphasis' 'Low latency IIR with de-emphasis' 'High attenuation with de-emphasis' 'Fixed process flow' 'Ringing-less low latency FIR'
  Item0: 'Ringing-less low latency FIR'
Simple mixer control 'Analogue',0
  Capabilities: pvolume
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 1
  Mono:
  Front Left: Playback 1 [100%] [0.00dB]
  Front Right: Playback 1 [100%] [0.00dB]
Simple mixer control 'Analogue Playback Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1
  Front Left: 0 [0%] [0.00dB]
  Front Right: 0 [0%] [0.00dB]
...

This card has multiple controls but we want to find a mixer control listed with a pvolume (playback) capability - in this case that mixer value required for the server configuration is called Analogue.

For the server configuration, we will use:

audio {
    nickname = "Computer"
    type = "alsa"
    card="hw:1"
    mixer="Analogue"
    # mixer_device=TBD
}

Mixer device

This is the name of the underlying physical device used for the mixer - it is typically the same value as the value of card in which case a value is not required by the server configuration. An example of when you want to change explicitly configure this is if you need to use a dmix device (see below).

Handling Devices that cannot concurrently play multiple audio streams

Some devices such as various RPI DAC boards (IQaudio DAC, Allo Boss DAC...) cannot have multiple streams openned at the same time/cannot play multiple sound files at the same time. This results in Device or resource busy errors. You can confirm if your sound card has this problem by using the example below once have determined the names/cards information as above.

Using our hw:1 device we try:

# generate some audio
$ sox -n -c 2 -r 44100 -b 16 -C 128 /tmp/sine441.wav synth 30 sin 500-100 fade h 0.2 30 0.2

# attempt to play 2 files at the same time
$ aplay -v -Dhw:1 /tmp/sine441.wav &
Playing WAVE '/tmp/sine441.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Hardware PCM card 1 'IQaudIODAC' device 0 subdevice 0
Its setup is:
  stream       : PLAYBACK
  access       : RW_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 44100
  exact rate   : 44100 (44100/1)
  msbits       : 16
  buffer_size  : 22052
  period_size  : 5513
  period_time  : 125011
  tstamp_mode  : NONE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 5513
  period_event : 0
  start_threshold  : 22052
  stop_threshold   : 22052
  silence_threshold: 0
  silence_size : 0
  boundary     : 1445199872
  appl_ptr     : 0
  hw_ptr       : 0
$ aplay -v -Dhw:1 /tmp/sine441.wav
aplay: main:788: audio open error: Device or resource busy

In this instance this device cannot open multiple streams - forked-daapd can handle this situation transparently with some audio being truncated from the end of the current file as the server prepares to play the following track. If this handling is causing you problems you may wish to use ALSA's dmix functionally which provides a software mixing module. We will need to define a dmix component and configure the server to use that as it's sound card.

The downside to the dmix approach will be the need to fix a samplerate (48000 being the default) for this software mixing module meaning any files that have a mismatched samplerate will be resampled.

ALSA dmix configuration/setup

A dmix device can be defined in /etc/asound.conf or ~/.asoundrc for the same user running forked-daapd. We will need to know the underlying physical soundcard to be used: in our examples above, hw:1,0 / card 1, device 0 representing our IQaudIODAC as per output of aplay -l. We also take the buffer_size and period_size from the output of playing a sound file via aplay -v.

# use 'dac' as the name of the device: "aplay -Ddac ...."
pcm.!dac {
    type plug
    slave.pcm "dmixer"
    hint.description "IQAudio DAC s/w dmix enabled device"
}

pcm.dmixer  {
    type dmix
    ipc_key 1024             # need to be uniq value
    ipc_key_add_uid false    # multiple concurrent different users
    ipc_perm 0666            # multi-user sharing permissions

    slave {
	pcm "hw:1,0"         # points at the underlying device - could also simply be hw:1
	period_time 0
	period_size 4096     # from the output of aplay -v
	buffer_size 22052    # from the output of aplay -v
	rate 44100           # locked in sample rate for resampling on dmix device
    }
    hint.description "IQAudio DAC s/w dmix device"
}

ctl.dmixer {
    type hw
    card 1                  # underlying device
    device 0
}

Running aplay -L we will see our newly defined devices dac and dmixer

$ aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
dac
    IQAudio DAC s/w dmix enabled device
dmixer
    IQAudio DAC s/w dmix device
default:CARD=ALSA
    bcm2835 ALSA, bcm2835 ALSA
    Default Audio Device
...

At this point we are able to rerun the concurrent aplay commands (adding -Ddac to specify the playback device to use) to verify ALSA configuration.

If there is only one card on the machine you may use pcm.!default instead of pcm.!dac - there is less configuration to be done later since many ALSA applications will use the device called default by default. Furthermore on RPI you can explicitly disable the onboard sound card to leave us with only the IQaudIODAC board enabled (won't affect HDMI sound output) by commenting out #dtparam=audio=on in /boot/config.txt and rebooting.

forked-daapd config with dmix

We will use the newly defined card named dac which uses the underlying hw:1 device as per /etc/asound.conf or ~/.asoundrc configuration. Note that the mixer_device is now required and must refer to the real device (see the slave.pcm value) and not the named device (ie dac) that we created and are using for the card configuration value.

For the final server configuration, we will use:

audio {
    nickname = "Computer"
    type = "alsa"
    card="dac"
    mixer="Analogue"
    mixer_device="hw:1"
}

Setting up an Audio Equalizer

There exists an ALSA equalizer plugin. On debian (incl Raspberry Pi) systems you can install this plugin by apt install libasound2-plugin-equal; this is not currently available on Fedora (FC31) but can be easily built from source after installing the dependant ladspa package.

Once installed the user must setup a virtual device and use this device in the server configuration.

If you wish to use your hw:0 device for output:

# /etc/asound.conf
ctl.equal {
  type equal;

  # library /usr/lib64/ladspa/caps.so
}

pcm.equal {
  type plug;
  slave.pcm {
      type equal;

      ## must be plughw:x,y and not hw:x,y
      slave.pcm "plughw:0,0";

      # library /usr/lib64/ladspa/caps.so
  }
  hint.description "equalised device"
}

and in forked-daapd.conf

alsa "equal" {
    nickname = "Equalised Output"
    # adjust accordingly for mixer with pvolume capability
    mixer = "PCM"
    mixer_device = "hw:0"
}

Using the web UI and on the outputs selection you should see an output called Equalised Output which you should select and set the volume.

When starting playback for any audio tracks you should hopefully hear the output. In a terminal, run alsamixer -Dequal and you'll see the eqaliser - to test that this is all working, go and drop the upper frequencies and boosting the bass frequencies and give it a second - if this changes the sound profile from your speakers, well done, its done and you can adjust the equalizer as you desire.

Note however, the equalizer appears to require a plughw device which means you cannnot use this equalizer with a dmix output chain.

Troubleshooting

  • Errors in log Failed to open configured mixer element when selecting output device
  • Errors in log Invalid CTL or Failed to attach mixer when playing/adjusting volumne mixer value is wrong. Verify name of mixer value in server config against the names from all devices capable of playback using amixer -c <card number>. Assume the device is card 1:
(IFS=$'\n'
 CARD=1
 for i in $(amixer -c ${CARD} scontrols | awk -F\' '{ print $2 }'); do 
   amixer -c ${CARD} sget "$i" | grep Capabilities | grep -q pvolume && echo $i
   done
)

Look at the names output and choose the one that fits. The outputs can be something like:

# laptop
Master
Headphone
Speaker
PCM
Mic
Beep

# RPI with no additional DAC, card = 0
PCM

# RPI with additional DAC hat (IQAudioDAC, using a pcm512x chip)
Analogue
Digital
  • No sound during playback - valid mixer/verified by aplay Check that the mixer is not muted or volume set to 0. Using the value of mixer as per server config and unmute or set volume to max. Assume the device is card 1 and mixer = Analogue:
amixer -c 1 set Analogue unmute  ## some mixers can not be muted resulting in "invalid command"
amixer -c 1 set Analogue 100%

An example of a device with volume turned all the way down - notice the Playback values are 0[0%]`:

Simple mixer control 'Analogue',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback 0 - 1
Mono:
Front Left: Playback 0 [0%] [-6.00dB]
Front Right: Playback 0 [0%] [-6.00dB]
  • Server stops playing after moving to new track in paly queue, Error in log Could not open playback device The log contains these log lines:
[2019-06-19 20:52:51] [  LOG]   laudio: open '/dev/snd/pcmC0D0p' failed (-16)[2019-06-19 20:52:51] [  LOG]   laudio: Could not open playback device: Device or resource busy
[2019-06-19 20:52:51] [  LOG]   laudio: Device 'hw' does not support quality (48000/16/2), falling back to default
[2019-06-19 20:52:51] [  LOG]   laudio: open '/dev/snd/pcmC0D0p' failed (-16)[2019-06-19 20:52:51] [  LOG]   laudio: Could not open playback device: Device or resource busy
[2019-06-19 20:52:51] [  LOG]   laudio: ALSA device failed setting fallback quality[2019-06-19 20:52:51] [  LOG]   player: The ALSA device 'Computer' FAILED

If you have a RPI with a DAC hat with a pcm512x chip will affect you. This is because the server wants to open the audio device for the next audio track whilst current track is still playing but the hardware does not allow this - see the comments above regarding determining concurrrent playback.

This error will occur for output hardware that do not support concurrent device open and the server plays 2 files of different bitrate (44.1khz and 48khz) back to back.

If you observe the error, you will need to use the dmix configuration as mentioned above.