Dear community,
- Create a playlist of a 24bps-96kHz track and a 16bps-44.1kHz track.
- Play the playlist without downsampling the 96kHz track.
Gapless playback does not work.
How can I solve this problem? Can someone please help me?
Best regards
Koray
Further information:
- In order to avoid the downsampling, I use ASIO output.
- I do not use WASAPI, because I have to change the sample rate from 96kHz to 44.1kHz between tracks manually, which is not gapless playback in the first place.
I found a solution, in which I use a resampler (dBpoweramp/SSRC) to upsample the 44.1kHz tracks.
However, I am still curious about other ideas or comments. It somehow bothers me, that the gapless playback fails. Isn't it a big deal?
Because of how soundcards usually work, I don't think it's possible to have gapless and glitchless playback while switching between different sample rates. It might be possible to minimize this problem to some extent, but it's probably not worth it, because it's an edge case.
The simplest solution is to keep the soundcard at a fixed sample rate and resample in the media player or the OS, as you also figured out.
I think it has nothing to do with Foobar2000.
The DA converter must change its clock from 44.1kHz to 96kHz or vice versa. There is no way that will be seamless unless your converter has an onboard DSP chip for such functions
8 (PRISM, Merging, Lynx, and Metric Halo, are some of the few that have that capability). Otherwise, you would need to have some intermediary DSP that makes everything you play the same sample rate.
If I misunderstand your question, please forgive me.
*For those converters, a part of their internal processing engine can convert sample rates in real time. This is a hardware-based version of some of the DSP modules in Foobar2000 that would do a similar function. - the thing in common is the converter's clock does not change its sample rate - the data is changed to fit the sample rate of the converter's clock.
Thank you for the discussion,
@Brand and
@Treelady !