Converting DSD to Flac?

[Spoon]

Re: Converting DSD to Flac?

1. 88KHz is a non standard audio frequency, not everything plays it in comparison to 96KHz
2. You would choose the option in dBpoweramp Control Center: Audio Codecs 'List / Options'

DSD Decoder >> Decoding Down Sampling

3. It is possible it will be looked at in the future, I am not aware of any complaints for the quality of DSD decoding.

[DigitalLA]

Re: Converting DSD to Flac?

in truth it is fs/2 (nyquist theorem)

With source DSD64 and target fs at 44.1kHz or 48kHz there is no need for a seperate lowpass filter as the signal is steep lowpassed at fs/2 in the SRC process.

For target fs 88.2kHz and above a steep lowpass filter starting at 30kHz in front of SRC is advisable. I did a little research and found above 26kHz you can't distinguish the music signal from the noise shaped noise, so I lowpassing at 26kHz with a steep filter min. 48dB/octave.

Advisable free lowpass filters are

- TDR Ultrasonic Filter (VST2 32 and 64 bit, alpha state but great working, parameters 25kHz, 100%, 100%)
- sox with: gain +6 sinc -L -26k, also good for SRC, -b 24 rate -v -a 88.2k dither (complete cmd: >source file< -b 24 <target file> gain +6 sinc -L -26k rate -v -a 88.2k dither)

If you don't want to burn a CD I would stay at 24bit. I'm usually doing DSD64 to PCM 24bit-88.2kHz.

Don't forget to dither to the final wordlength (16bit or 24bit) as final step.

G.

I just saw your response. Thank you.

Is it safe to assume that the record company that produced the original DSD, 96/24, or 192/24 files filtered out harmful frequencies? Otherwise, buyers could be subjecting themselves or others to potential harm just by playing what they purchased.

[hjam]

Re: Converting DSD to Flac?

1. 88KHz is a non standard audio frequency, not everything plays it in comparison to 96KHz
2. You would choose the option in dBpoweramp Control Center: Audio Codecs 'List / Options'

DSD Decoder >> Decoding Down Sampling

3. It is possible it will be looked at in the future, I am not aware of any complaints for the quality of DSD decoding.

A few years ago, I used dbPoweramp to convert several DSF files to 88khz pcm files. (I was following advice I saw elsewhere to stick with an even multiple of the DSD frequency - 88khz, 176khz, etc.). But I noticed they did not sound as good as some 96khz flac files of the same recordings that I acquired elsewhere. So, I used dbPoweramp once again to convert from DSF to 96khz 24 bit flacs. Better! To my ears, it seems like there's less of that edgy digital sound in the 96khz FLACs.

Spoon, I'm curious. Is there an obvious explanation? Or, is this Listener and Sound System dependent?

Thanks,

Hugh

[Spoon]

Re: Converting DSD to Flac?

Possibly your sound system is better at playing back standard frequencies, ie: 44.1KHz, 48KHz, 96KHz, 176KHz, 192KHz

[hjam]

Re: Converting DSD to Flac?

That sounds like a reasonable explanation. Maybe I'll try recording some LPs at 88khz and 96khz and see if I get the same thing on playback.
Thanks Spoon.
Hugh

[Brewbs]

Re: Converting DSD to Flac?

Actually, I agree with the 44.1/96/192 not being significant. But I don't agree with 16/24. The latter is to my ears an audible difference and an improvement. I have done comparisons between 44.1/16 and 44.1/24, as well as 96/24, and the difference betwenn the two 44.1 was much clearer than the 44.1 v 96.

That's a little outlandish. The extra dynamic range is very useful in the studio headroom and processing purposes but in home listening...well, no.. there's more capacity to provide deafening output. However if 16 bit material has been arrived at from a 24 bit source and no dithering has been applied, then artifacts will likely have been introduced. The other concern with 16 bit is by using a digital volume control you can reduce the bit depth, a lot of people write this off as imperceptible but believe me, combine highly sensitive speakers with a very healthy amplifier wattage and a digital volume control (I did this with my Klipsch horns and PS Audio mono blocks) and it is quite easy to degrade the audio output to an unlistenable level. So that's the only limited circumstance where a higher bitrate could help when playing digital files, but it can be easily averted by paying attention to your signal chain.

My focus is usually on limiting bitrate and frequency where possible to avoid intermodulation distortion or any other reproduction issues. Most gear can't even reproduce a lot of this "hi res" music. But don't listen to me, there are plenty of other engineers out there who have debunked this and some of the more recent ones have helpfully focused on how it will often be detrimental to audio quality.