Re: The rate parameter in asound.conf

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Jeffrey Barish wrote:
> Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 88200 Hz, Stereo
> Warning: rate is not accurate (requested = 88200Hz, got = 96000Hz)
>          please, try the plug plugin
> Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0
>
> When I try the command with plughw, I get
>
> aplay -D plughw:0 -t raw -f dat -r 88200 -d 3 -v /dev/zero
> Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 88200 Hz, Stereo
> Plug PCM: Rate conversion PCM (96000, sformat=S16_LE)
> Converter: linear-interpolation
> Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0
>
> If I have this right, when I use plughw, ALSA is performing the sample rate
> conversion;

Yes.

> otherwise, it is the DAC itself.

The DAC never converts the sample rate; it just runs at that rate.

> When I specify any other sample rate, the sample rate of the DAC is the same
> as the sample rate specified in the aplay command.  Can I assume that ALSA is
> doing nothing to the signal in those cases?

Yes.

> By the way, the DAC is Realtek ALC892.

This chip supports 88.2 kHz only on the digital output.

> When I run ALSA with plughw and ALSA invokes the sample-rate converter to go
> from 88200 to 96000, I see from the output above that it uses linear
> interpolation.  Is there a way to specify a better algorithm?

Install the alsa-plugins package.  (Which might have been split into several
package by your distribution.)


Regards,
Clemens

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