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Recorded in 16bit/44.1 KHZ but playback in 24bit/96khz??

post #1 of 76
Thread Starter 

Hey there - question:

 

If my ALAC files are recorded in 16bit/44khz - is it better to playback at 24bit/96khz or do I need to match the recorded bit/sample rate??

 

Because my DAC can playback at the the 24bit/96 khz - I was thinking of making that my setting??

 

H

post #2 of 76
Thread Starter 

In fact - I did play it back at 24bit/96KHZ and I got lots of crackling sounds which I assume is coming from the DAC?? Any one know about this issue? I am using the Yulong D100...and my ALAC files are at 16bit/44.1 KHZ

post #3 of 76

Dont resample. Ever.

 

The crackling is most likely coming from the DAC trying to upconvert the digital signal.

post #4 of 76

Resampling just screws up the sound.

 

I don't know about that DAC, though.

post #5 of 76

I have no problem playing files originally at 16/44  up-sampled to 24/96. Up-sampling, I've been told occurs in Windows. My route is USB from Dell Inspiron 14 to X-FI HD, and then, optical out from X-FI HD to Sony TA-E9000ES optical input, which will receive a 24/96 signal. Perhaps, your crackle has something to do with buffering.

post #6 of 76
Thread Starter 

So in other words - the setting in Quick time - for itunes should stay 16bit/44.1KHZ. I thought the issue was the Quick time program trying to do the resampling rather than the DAC itself...

post #7 of 76
Thread Starter 


If it is buffering - where do I alter the setting? Another question - I thought that itunes/quick time was doing the upsampling - but surely I should leave those settings at the original ALAC file level of 16 bit/44.1 khz and then let the DAC do the upsampling - if that is possible?

 

Also - do I need to alter both the sample rate conversion and the bit rate or which one? Sorry - I'm new at computer audio...

 

H

 

Quote:
Originally Posted by sterling1 View Post

I have no problem playing files originally at 16/44  up-sampled to 24/96. Up-sampling, I've been told occurs in Windows. My route is USB from Dell Inspiron 14 to X-FI HD, and then, optical out from X-FI HD to Sony TA-E9000ES optical input, which will receive a 24/96 signal. Perhaps, your crackle has something to do with buffering.

post #8 of 76

I've heard that say Foobar2k upsamples everything to 24bit regardless?

post #9 of 76
Quote:
Originally Posted by Mr Pink57 View Post

I've heard that say Foobar2k upsamples everything to 24bit regardless?



Resampling only occurs when a resampler is in the DSP chain. WASAPI, ASIO or KS do warrant bit-matched playback, so no resampling occur on either the player or the hardware through windows settings.

post #10 of 76
Quote:
Originally Posted by Mr Pink57 View Post

I've heard that say Foobar2k upsamples everything to 24bit regardless?


You can set it to output 24-bit depth, but no upsampling occurs, it simply adds zeroes to the signal.  In other words, it does nothing, but then if you play some 24-bit music, it will not get reduced to 16-bit depth.

Upsampling/downsampling/resampling is done through the PPHS resampler or the sox resampler.  They will up-/down-sample the music's sample rate to 32kHz, 44.1kHz, 48kHz, 88.1kHz, and 96kHz (and more but those are the main ones most gear will take in one way or another).

post #11 of 76
Thread Starter 

How to later the latency /buffer size on itunes?? 

post #12 of 76

You can set the buffering in the Quicktime Control panel under the Audio tab.  You set the latency to a higher or lower number, whatever ends up working out best for you.

post #13 of 76
Thread Starter 

the default setting is 33.000ms - I have heard that 1000 ms is supposed to be right. It is really strange - after 5 or 10 minutes playback - popping starts in the audio...

post #14 of 76

You won't damage your gear with incorrect latency settings.  Try different numbers, gradually incrementing to 1000ms until you find what works out best.

I did the same when I was starting to use Foobar2k + ASIO.

post #15 of 76
Thread Starter 

Cheers! so you had popping/cracking issues with USB playback? Or was it an issue of upsampling? Just worried that my Yulong D100 is the problem/

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