Recorded in 16bit/44.1 KHZ but playback in 24bit/96khz??
Feb 22, 2011 at 9:42 AM Thread Starter Post #1 of 76

Syan25

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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
 
Feb 22, 2011 at 12:19 PM Post #2 of 76
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
 
Feb 22, 2011 at 4:17 PM Post #4 of 76
Resampling just screws up the sound.
 
I don't know about that DAC, though.
 
Feb 22, 2011 at 4:24 PM 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.
 
Feb 22, 2011 at 5:45 PM Post #6 of 76
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...
 
Feb 22, 2011 at 5:46 PM Post #7 of 76

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:
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.



 
Feb 22, 2011 at 9:02 PM Post #8 of 76
I've heard that say Foobar2k upsamples everything to 24bit regardless?
 
Feb 22, 2011 at 9:48 PM Post #9 of 76


Quote:
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.
 
Feb 22, 2011 at 11:55 PM Post #10 of 76


Quote:
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).
 
Feb 23, 2011 at 12:45 AM 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.
 
Feb 23, 2011 at 12:47 AM Post #13 of 76
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...
 
Feb 23, 2011 at 12:50 AM 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.
 
Feb 23, 2011 at 1:00 AM Post #15 of 76
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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