What happens when 44.1 plays at 192khz?

[Wombat]

After reading that i wonder what credit such posts expect.
http://www.diyaudio.com/forums/pc-based/232019-linux-audio-way-go-no-its-not-12.html#post3603842

 

[phofman]

This also explains why my jackd usb data through my usb to spdif sounded worse than my cd player's spdif output straight to my dac (resampling in jack is probably responsible).

Your CD spdif output is a different hardware than your USB soundcard with SPDIF output. It is perfectly possible they sound differently, without any resampling (different jitter, different level of noise into the SPDIF receiver). Only a blind test could confirm that, of course.

But IMO the problem is you do not even know if your PC setup is resampling or not. Why do we actually discuss here esoteric issues like latency effect when such basic stuff is still unknown.

Again, please learn how your setup works first. My best advice I can give, honestly.

 

[jkorten]

Your CD spdif output is a different hardware than your USB soundcard with SPDIF output. It is perfectly possible they sound differently, without any resampling (different jitter, different level of noise into the SPDIF receiver). Only a blind test could confirm that, of course.

But IMO the problem is you do not even know if your PC setup is resampling or not. Why do we actually discuss here esoteric issues like latency effect when such basic stuff is still unknown.

Again, please learn how your setup works first. My best advice I can give, honestly.

Audiophiles don't use sound cards. They use only expensive outboard USB to S/PDIF converters that cost anywhere from $200 to $5000. (Those low jitter crystals are expensive you know.)

I am assuming that playback application using gstreamer to ALSA is not resampling if the USB->S/PDIF is capable of the native sample rate. Without reading the source code, I can only make this assumption.

Jackd appears to be something audio production people use to route between apps, and also appears to be optimized for a single sample rate that is shared among all the applications (no resampling). It is strange how it is advocated for use during "bit perfect" audio playback at the ap-linux website. The fact that it represents everything as floating point doesn't bother me as much, as rounding back to integer (16 or 24 bit) will still be an order of magnitude less precision than the floating point.

 

[phofman]

Audiophiles don't use sound cards. They use only expensive outboard USB to S/PDIF converters that cost anywhere from $200 to $5000.

I understand the Nevertheless, anything which is detected as a sound device is a soundcard, be it an integrated soundchip or USB soundcard with SPDIF ouput only.

I am assuming that playback application using gstreamer to ALSA is not resampling if the USB->S/PDIF is capable of the native sample rate.

It depends what output device is configured in alsasink of gstreamer. If no specific device is configured by the calling application, it takes the default device

gstreamer/gst-plugins-base - 'Base' GStreamer plugins and helper libraries

gstreamer/gst-plugins-base - 'Base' GStreamer plugins and helper libraries

The default device is configured to go through dmix (or pulseaudio) on stock alsa installations.

Jackd appears to be something audio production people use to route between apps, and also appears to be optimized for a single sample rate that is shared among all the applications (no resampling). It is strange how it is advocated for use during "bit perfect" audio playback at the ap-linux website.

Why should it surprise you