Troubleshooting

I just updated Raspotify and now none of my settings from /etc/default/raspotify work anymore!

Version 0.31.4 introduces breaking changes to the Raspotify Package.

Please see the wiki for details.

My volume on Spotify is 100% and it's still too quiet!

Have you tried turning the volume up using the command alsamixer?

My Raspberry Pi does not use my sound card/device!

Well 1st let's find your card/device (DAC) with aplay.

aplay -l should output something that looks like this:

**** List of PLAYBACK Hardware Devices ****
card 0: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 2: DAC [JDS Labs Element DAC], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 2: DAC [JDS Labs Element DAC], device 1: USB Audio [USB Audio #1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Now edit /etc/asound.conf. (It may or may not already exist.)

Add:

defaults.ctl.card <your DAC's card #>
defaults.pcm.card <your DAC's card #>

In the example above the DAC's card # is 2 so:

defaults.ctl.card 2
defaults.pcm.card 2

Now you can test it out with:

speaker-test -c2 -l1

If you hear noise you're good to go. Your DAC is configured as the default sound device.

I'm an audiophile and I want bit-perfect output!!!

Well, sorry... Spotify doesn't stream lossless audio currently.

To get as close as possible we're going to need to know what format(s) your DAC supports.

Wait, isn't the audio 16 bit 44.1 khz?

Well, it started out that way but lossy audio doesn't have a bit depth.

The decoder outputs 32 bit float 44.1 khz PCM and all digital processing in librespot is done in 64 bit float to be as lossless as possible.

That PCM has to be re-quantized back into an integer format for playback on most DACs.

The higher the bit depth the lower the quantization noise.

So we want librespot to output the highest bit-depth your DAC supports to get the highest quality sound possible.

To find out our example DAC's supported formats we're going to take the card # from above and use aplay again:

aplay -Dhw:2 --dump-hw-params /usr/share/sounds/alsa/Front_Right.wav

That should output something that looks kinda like this:

ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT:  S16_LE S24_3LE S24_LE S32_LE
SUBFORMAT:  STD
SAMPLE_BITS: [16 32]
FRAME_BITS: [32 64]
CHANNELS: 2
RATE: [8000 192000]
PERIOD_TIME: (166 8192000]
PERIOD_SIZE: [32 65536]
PERIOD_BYTES: [256 524288]
PERIODS: [2 4096]
BUFFER_TIME: (333 16384000]
BUFFER_SIZE: [64 131072]
BUFFER_BYTES: [256 524288]
TICK_TIME: ALL
--------------------

What we care about is FORMAT and RATE.

If the device supports a sampling rate (RATE) of 44100 we want to use that to avoid resampling. If not it will more than likely support 48000.

The example card does support 44100. ([8000 192000] is a range)

So in our /etc/asound.conf we want to add:

defaults.pcm.dmix.rate 44100
defaults.pcm.dmix.format S32_LE

All together our example /etc/asound.conf looks like this:

defaults.ctl.card 2
defaults.pcm.card 2
defaults.pcm.dmix.rate 44100
defaults.pcm.dmix.format S32_LE

librespot doesn't care about the endianness (the LE or BE at the end) it will take care of that for us.

To set the format to S32_LE change:

# Output format {F64|F32|S32|S24|S24_3|S16}. Defaults to S16.
#LIBRESPOT_FORMAT="S16"

To:

# Output format {F64|F32|S32|S24|S24_3|S16}. Defaults to S16.
LIBRESPOT_FORMAT="S32"

If you want to get really crazy you can disable volume normalization and volume control and set the initial volume to 100 to approximate bit-perfect as much as possible by uncommenting:

# Initial volume in % from 0 - 100.
# Defaults to 50 For the alsa mixer: the current volume.
#LIBRESPOT_INITIAL_VOLUME="50"

# Volume control scale type {cubic|fixed|linear|log}.
# Defaults to log.
#LIBRESPOT_VOLUME_CTRL="log"

And changing it to:

# Initial volume in % from 0 - 100.
# Defaults to 50 For the alsa mixer: the current volume.
LIBRESPOT_INITIAL_VOLUME="100"

# Volume control scale type {cubic|fixed|linear|log}.
# Defaults to log.
LIBRESPOT_VOLUME_CTRL="fixed"

The audio output buzzes a few seconds after audio stops!

This is likely to be ALSA's Dynamic Audio Power Management (DAPM) shutting down the sound module of your device to save power.

If you want to disable this feature, create a file called snd_soc_core.conf in /etc/modprobe.d with this line in:

options snd_soc_core pmdown_time -1

Once you reboot and play some sound, the issue should be gone.

How do I use/enable hardware volume control in librespot?

The short answer is you don't if you care about fidelity.

Unless you need some kind of advanced ALSA feature that requires you to use hardware volume control don't use it.

Software volume control in librespot in done with 64 bit floating point precision and is objectively better than any and all forms of hardware volume control.

I'm using the pipe backend with snapcast / a fifo and the raspotify systemd service keeps crashing!

The provided service file uses best practices to limit the access that raspotify has to various system directories as much as possible.

Avoid using /tmp/ for your FIFO file. Instead place it somewhere under /var/run/.

Other issues

File an issue and if we get it sorted, I'll add to this list.