Resampling of Shairport-Sync output

[MKSounds]

Hello,

I'm using the I²S output of a Raspberry Pi 2b with Volumio 2.389 to provide audio data to a dsp (ADAU1701). The dsp must be configured as clock-master (so the raspberry is a clock slave) and because it doesn't have an ASRC, the input sample rate has to be fixed at 48 kHz (because of the mounted oscillator). I use my overlay in combination with the onboard resampling of volumio. In this case everything works perfect with local media.

Now I want to use AirPlay to stream my audio to volumio, but the output of shairport-sync is limited to 44,1 kHz and it's multiples. Then I realised, that the resampling doesn't affect the output of AirPlay content, which is really annoying. The result is a badly stuttering audio playback (1 second playing as it should, 2 s break, 1 s playing...).

So the question is: what is the easiest way to resample the content from the shairport-sync output as well ?

Best regards, Markus.

[asvedulf]

Any news on this?

[timothyjward]

As of Volumio 3 shairport audio is now treated the same way as other audio. If you write a simple Volumio plugin to force 48kHz resampling then it should work for all audio sources including Shairport.

There is documentation about how to plug into AAMPP here

[asvedulf]

Ok, might need some help here.

Should I disable the Settings -> Playback Options -> Audio Resampling?

I have created a plugin with has_alsa_contribution: true. File with below contents -> plugin_dir/asound/resampler.Slave.conf:

pcm.resampler {
        type plug
        slave {
                pcm "Slave"
                format S24_LE
                rate 48000
        }
}
pcm.!default resampler

Double checked that the plugin is loaded in dev mode in live log. Name of Soundcard is "Slave" referring to Card 1, Subdevice 0. Have RPI Zero W connected to ADAU1701 over I2S. Using the internal clocks of the ADAU1701. Output sound is distorted.

[timothyjward]

Should I disable the Settings -> Playback Options -> Audio Resampling?

It’s not required that you do, but it’s recommended. If you put different settings in your audio will get resampled twice.

Name of Soundcard is "Slave" referring to Card 1, Subdevice 0.

Don’t target the sound card. As per the documentation you should target a pcm name that doesn’t exist so that Volumio can create it for you and join your contribution to the next stage in the AAMPP.

Have RPI Zero W connected to ADAU1701 over I2S. Using the internal clocks of the ADAU1701. Output sound is distorted.

Can you see your contribution in /etc/asound.conf?

If so, Is it distorted with all audio, including native 48kHz? If so it’s likely a driver/hw issue. If it works fine with a 48kHz input file then it will be the resampling.

You should be able to see what rate is being used with your device by using cat to display the audio device information under /proc

[volumio]

@timothyjward @mervinderuiter

This global resampling feature should be part of core Volumio. Who can have a look at it?

[asvedulf]

Got it working according to AAMPP with below snippet. /proc device shows 48000 Hz. /etc/asound.conf adds an entry as well.

pcm.adau_in {
        type plug
        slave {
                pcm "adau_out"
                format S24_LE
                rate 48000
        }
}

Sound quality is better than comparing with my separate bluetooth dongle.

But agree with above, this alsa-configuration possibility should be part of core system.