[zilch0md]

The Sony PCM-M10 LCPM flash deck is capable of playing 96-kHz/24-bit WAV files.   I've been downloading 96-Khz/24-bit FLACs from HDtracks, then decompressing them to WAV using dbPoweramp, but leaving them at 96/24. 
 
Obviously, there are a lot more recordings available in 44.1-kHz/16-bit CD format, which leads to my question:

In your opinion, would it be best, in terms of SQ, to just convert 44.1/16 content to WAV for playback on the PCM-M10, without upsampling or increasing the bit depth to 96/24?

I'm eager to learn so feel free to correct me if I say something ignorant here, but at the moment, I believe that a bit depth of 24-bit offers nothing over 16-bit in terms of improved SQ on playback, because the dynamic range supported by 16-bit is more than sufficient for both our ears and our audio hardware.  (I know, however, that even 32-bit can be of great benefit when recording.)

My question is more to the subject of sampling frequency - specifically, whether or not there's any benefit to be enjoyed (on playback), having upsampled a 44.1-kHz file to 88.2- or 96-kHz.

Which leads to another question:  The specs for the PCM-M10 make no mention (that I've seen) of support for 88.2 kHz sampling.  Does anyone know if it can play an 88.2 kHz WAV, even though it can't record at that frequency?  If so, does anyone believe that upsampling from 44.1-kHz to 88.2 kHz would provide superior results on playback vs. upsampling from 44.1 to 96?  And, if so, can you elaborate or debunk the oft' heard reasoning that a 2x multiplier just has to sound better than a 2.18x multiplier?

Lastly, to save space (33%) on the microSDHC cards, I'm thinking of using dbPoweramp to reduce the bit depth of the 96/24 content purchased online to 96/16 (without changing the sampling frequency).   Does anyone think, again in terms of SQ on playback, that I'd be better off leaving those files alone (as 96/24 WAVs)?  Not so much to preserve the 24-bit dynamics, which I contend cannot be appreciated, but rather to simply avoid a possible loss in playback quality caused by "over-processing" the files - in adding one more step to the workflow.

Thank you everyone, in advance.  I'm looking forward to your responses.

Mike  

 

[Anaxilus]

Playing an unspecified format like 24/88.2 may or may not work, try it.  
 
You realize upsampling from 16/44 to 24/96 will do nothing right?  You are not gaining any new data because there is nothing there to be gained.

 

[nikongod]

I think this is a good place to ask:
 
Does anyone know of anything that will record and play 2 channel 24/96 from a digital input & output on the cheap?
 
Im looking for a 2496 Iriver IHP120, basically. I found a sony thing on that link that looks neat, would rather not pay $450. I looked at a few of the ZOOM products as well, but none have digital in/out.  
 
It can not be PC based unless it runs on Linux.

 

[Anaxilus]

Quote:

I think this is a good place to ask:
 
Does anyone know of anything that will record and play 2 channel 24/96 from a digital input & output on the cheap?
 
Im looking for a 2496 Iriver IHP120, basically. I found a sony thing on that link that looks neat, would rather not pay $450. I looked at a few of the ZOOM products as well, but none have digital in/out.  
 
It can not be PC based unless it runs on Linux.

I would shoot LFF a PM on that.  Out of my depth on a digital HD recorder.

 

[zilch0md]

Thanks for the reply!

I'm right with you on the understanding that upsampling cannot add information that wasn't there to begin with, but I've been doing a lot of reading about thd filtering that is performed by DACs and although the finer points completely escape me in some of thr material I've studied, I'm certain threre are some smart people out there who believe there is a lot to be gained in the accuracy of playback (and lack of distortion).

This makes sense when you consider that a processor fast enough to perform DAC functions with recordings that were ADC'd at 96-kHz will be handling a 44.1-kHz stream twice as fast as a 44.1-kHz DAC would - if and only if that 44.1-kHz data were disguised as 96-kHz data.

One guy put it like this (paraphrasing): "If you are translating a foreign language to English, would't it be great if the person you are interpreting spoke at half the speed you're comfortable with? You're translation would be far more accurate."

If you've heard arguments like this before and can point me to a good reference that disputes this position, I'm still educatable.

What do think about reducing the file size of 96/24 recordings (obtained from Linn or HDtracks) by 33% using dbPoweramp to reduce the bit depth from 24 to 16? As I said earlier, I don't believe the loss of dynamic range can be detected audibly, but I'm concerned about over-processing the file. Do you have any insights to offer on this? (Again, I am a lump of clay at this point - willing to be taught if the argument makes sense.)

Thanks,

Mike

Playing an unspecified format like 24/88.2 may or may not work, try it.  


 


You realize upsampling from 16/44 to 24/96 will do nothing right?  You are not gaining any new data because there is nothing there to be gained.

 

 

[zilch0md]

Hey Nikongod,

You might try posting your question at www.taperssection.com, a forum for people who do audio recording. Registration is fast and they are a friendly bunch - most of them are musicians.

Mike

 

[anadin]

Try listening yourself then decide.

 

[zilch0md]

Quote:

Try listening yourself then decide.

With a handful of tracks that I've tested, I can't tell any difference between 44.1 kHz/16-bit and an upsampled version of the same file at 96-kHz/24-bit, but my inability, thus far, to hear a difference using those particular files, with my ears, my processes, and my hardware, does not mean there is no difference. 

 

[zilch0md]

Thanks for all the feedback. I have concluded that the majority believes upsampling accomlishes nothing, so that's the position I have accepted. I did my best to present the opposing minority argument, but it seems no one is here to defend it. I'm no longer on the fence.

 

[3602]

I thought the whole point of "upsampling" is so that the DAC noise doesn't reach the analog stage, and upsampling makes it easier to impose a less evil filter.
Or am I confounding that with "oversampling". Hmm. Collecting vintage PCDP's does things to you.

 

[GeorgeGoodman]

Do a few more A/B tests with a friend, and tell him or her to not let you know which version they are playing. Also, what headphones/amps/other equipment are you using for the comparisons? If your system is not resolving enough, then you may have a very hard time discerning a difference in sound. Having said that, I know I can hear some pretty small differences in quality in a blind test using pretty low grade equipment like stock earbuds and mac computer speakers. Do a few more tests and see what happens. I definitely agree though, that well mastered 16/44 track will sound better than a 24/96 track not mastered as well. Don't sweat the differences too much and just enjoy the music. If you find you really can;t hear a difference on decent equipment, move on and get some well mastered music or upgrade your system somewhere else. 16/44 can sound amazing with the right DAC, amp, headphones etc. Heck, you can skip the DAC and amp and still get amazing sound with the right headphones.

 

[3602]

Here's something people should try to do. Record at the same time, the same sound source, with two identical setups as close together as possible, with one setup at 16/44 and the other at 24/96. This way you'd get (hopefully) identical recordings, but one with higher resolution than the other. Then try do distinguish one from the other. But for this to work the two mikes need to be infinitely close, so...
Or, you can get a real piece of 24/96 and dither/downsample it to 16/44 and see if you can tell. Hey I can't. Not even on my PCM-D50.