Samplerate change impossible, soundcard problem

Hi, I use ardour with manjaro and have problems adjusting samplerates. With Alsa I can only use 48000, other rates don’t work. With Jack, Ardour crashes. With pulseaudio other rates work, but there is no MIDI system selectable (only software keyboard).

I tried to adjust the rate of the hardware with arecord --rate=44100 without success (cryptical output, crash)
inxi -A:
Audio: Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel
Device-2: Roland Roland Digital Piano type: USB driver: snd-usb-audio
Sound Server-1: ALSA v: k5.9.16-1-MANJARO running: yes
Sound Server-2: PulseAudio v: 14.2 running: yes
Sound Server-3: PipeWire v: 0.3.23 running: yes
I know that the build-in soundcards are not the best, but different samplerates should be possible. Any suggestions?

Most likely your audio device only supports 48k. Given the list of devices I wouldn’t be at all surprised if this was the case, it is very common particularly with cheaper USB devices and built in audio devices.

Other options that allow you to use other sample rates are probably resampling to allow it.

  Seablade

Well, other samplerates run with pulseaudio (except the missing midi problem). So the soundcard should not be the problem.
Audio Adapter: Intel Kaby Lake-U/Y PCH - High Definition Audio Controller
Realtek ALC3253CG

Again see my last comment, Pulseaudio is likely resampling all inputs to 48k to match what your sound card is capable of. Jack and Ardour with ALSA won’t do that, they will keep it at the same sample rate your card is capable of so that there is no unnecessary resampling.

  Seablade

Pulseaudio is a desktop sound system. It is audio-only and does not support MIDI.
It is also unreliable and hence Ardour only supports playback via pulseaudio.

As @seablade mentioned pulseaudio will resample in software to match sample-rates.

Many Intel HDA devices only support 48KHz sample-rate.

1 Like

OK, I didn’t understand that Pulseaudio is resampling, thought that seablade meant ardour resamples. Not to believe that a 1000 bugs machine has only one samplerate and no working asio driver… Next PC is a chinese one.
Thank you

Try (in a terminal) less /proc/asound/PCH/codec#0
You will see a line that says something like:
Default PCM:
rates [0x5f0]: 32000 44100 48000 88200 96000 192000
Up near the top before any “Node” lines. Ignore this line as it is only the rates the PCH bus is able to handle not the codec. After that there are a bunch of Node groups that start with a Node line that has a name of the codec the indented part under refers to. The first one on my system is:
Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out
That is my Front left and right it shows:
rates [0x560]: 44100 48000 96000 192000
Note that it shows fewer rates than the default, yours may show only 48000 or 48000 and 96000 or something like that.

Note that 48000 is the “standard” since at least the 1970s. 44100 was only for CDs and nothing else. Microsoft uses 48000 internally for just about everything and resamples to 48000 internally for many things. The older AC97 internal sound devices were all 48000 codecs with a hardware Sample Rate Conversion in line to provide other rates (even 48000 itself in some cases). 44100 has been used by many people because: A) they ripped stuff from CDs or B) they were creating for CDs or C) they heard 44100 a lot and assumed it was the “standard”. So today 44100 is a standard of sorts even though CDs are quietly slipping away. There are even some audio devices that have 44100 only.