Now that I‘ve learned how to extract ISO / DSF from my SACDs thanks to another thread of this forum, I‘m looking for a way to convert the resulting files into a format my Rotel PreAmp can process when that tiny Daphile server sends the data. The Rotel RC-1570 is not capable of processing DSD, only PCM up to 192 kHz / 24 bit.
I‘m still rather new to this topic. When reading articles or posts about it, one claim appears to be popping up frequently: ‚Conversion from DSD to PCM is (always) a lossy process.‘
But is that really correct? I did a little bit of primary school math:
At 2822.4 kHz @ 1 bit, an SACD delivers a data stream of 2,822,400 bps. (To the DJs around: this is bits per second, not beats per second 😉 )
Resampled to 88.2 kHz @ 24 bit, which is often suggested for an SACD, our target files should deliver 2,116,800 bps. Well, this is lossy.
Choosing 176.4 kHz @ 16 bit instead would deliver an output stream of 2,822,400 bps, exactly like our SACD source. From a naive point of view, I would not consider this lossy.
If we now encapsulate our PCM file into a FLAC container, compressing it by lossless algorithms, where should losses of the original audio quality appear?
Looking at the topic this way, theoretically a resampling to 88.2 kHz @ 32 bit should also be lossless, only my Rotel cannot process this, and it seems to be rather uncommon. So 176.4 kHz @ 16 bit should be my format of choice, at least until I do not have a DAC that can process DSD directly.
Are these considerations correct or am I missing something?
Some sources suggest to choose always the highest quality the DAC in use can handle if converting from DSD to PCM. In my case, choosing 192 kHz @ 24 bit would increase the resulting file sizes by 63 %. Do I get some audio quality in return?
I‘m still rather new to this topic. When reading articles or posts about it, one claim appears to be popping up frequently: ‚Conversion from DSD to PCM is (always) a lossy process.‘
But is that really correct? I did a little bit of primary school math:
At 2822.4 kHz @ 1 bit, an SACD delivers a data stream of 2,822,400 bps. (To the DJs around: this is bits per second, not beats per second 😉 )
Resampled to 88.2 kHz @ 24 bit, which is often suggested for an SACD, our target files should deliver 2,116,800 bps. Well, this is lossy.
Choosing 176.4 kHz @ 16 bit instead would deliver an output stream of 2,822,400 bps, exactly like our SACD source. From a naive point of view, I would not consider this lossy.
If we now encapsulate our PCM file into a FLAC container, compressing it by lossless algorithms, where should losses of the original audio quality appear?
Looking at the topic this way, theoretically a resampling to 88.2 kHz @ 32 bit should also be lossless, only my Rotel cannot process this, and it seems to be rather uncommon. So 176.4 kHz @ 16 bit should be my format of choice, at least until I do not have a DAC that can process DSD directly.
Are these considerations correct or am I missing something?
Some sources suggest to choose always the highest quality the DAC in use can handle if converting from DSD to PCM. In my case, choosing 192 kHz @ 24 bit would increase the resulting file sizes by 63 %. Do I get some audio quality in return?