On Thursday, 08 May 2014 08:52:55 you wrote:
> 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
Thanks for your guidance on this problem.
For anyone who is interested, here is the asound.conf that seems to solve my
problem playing sound files with a sample rate of 88.2 kHz:
pcm.!default {
type plug
slave {
pcm hw
}
}
ctl.!default {
type hw
card 0
}
defaults.pcm.rate_converter "speexrate"
The computational burden of the speexrate converter is only 3% on my system.
By contrast, samplerate_best was 46%.
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