Jakob Hansen wrote:
> I have noticed a distinct degradation in audio quality when using the
> Master Playback volume control in amixer. Is this an artifact of
> truncation being done in the hardware
Maybe; this happens if you attenuate 16-bit samples too much.
Is that mixer control implemented in hardware, or emulated?
> and is it possible to use dmix or any other means to increase the bit
> depth before the signal reaches the dac?
dmix must use a sample format that is supported by the hardware.
> sometimes setting certain parameters in asound.conf can yield
> unexpected results, for example in the following pcm declaration taken
>> almost directly from the alsa documentation (this is just one block,
> not the entire file. )
>
> pcm.dmixer {
> type dmix
> ipc_key 1024
> slave {
> pcm "hw:1,0"
> format S16_LE
> period_time 0
> period_size 1024
> buffer_size 8192
> rate 44100
> }
>
> The rate is declared to be 44100, but when I issue
> 'cat /proc/asound/card1/pcm0p/sub0/hw_params', the rate appears to
> be 48k.
Apparently, your hardware does not support 44.1 kHz.
Why are you trying to force it to use that rate?
> The odd thing is that no matter what the input file format is, it
> plays back at the correct speed (or pitch level) whether the rate is
> declared to be 44.1k or 48k. But, if I declare the rate to be 96k,
> then hw_params shows 96k and the file plays back at double speed,
> which I suppose makes some kind of sense.
Do you have some "plug" plugin on top of the dmix plugin? Or are you
using a player program that can do its own resampling? If you have
neither, it's possible that your player just dumps 48 kHz samples into
the device.
> changing the format declaration apparently has no effect whatsoever.
What formats does your hardware support?
Regards,
Clemens
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