Originally Posted by ab initio
I have a nit to pick here.
This makes zero sense.
Upsampling involves changing the sampling rate of the audio. All you are doing by converting 16-bit to 24-bit audio is zero padding eight 0's to the end of your 16-bit PCM words.
My next question is this: why on earth do any of your CD's have a DC offset? Let's say you have a 60 second sample with approximately 20 Hz content at 0dB full scale. your "DC offset" by chance can never be greater than that from a single half-cycle of this 20Hz info averaged over the length of your sample. In this example here, the DC offset of this sample is 1/(60*20*pi).
If you compare that to 16bit depth, that DC offset works itself out to be 8.3 counts of audio. That's 8 out of the 65536 possible values of 16 bit audio!
When you subtract the DC offest do you ever have audio clips with a bigger DC offest than 8 out of 65536? If so,
what CD's are these and who mastered them, because I would really like to avoid buying anything from them in
I guess my point is this: You aren't worried about the "DC offest" during the first 1/40th of a second at the start of a
20Hz signal, so why would you care otherwise? Do you really have tracks that you listen to with enough extreme low
frequency content at significant amplitude such that there is appreciable "DC offset" at > O(1) sec? I guess I don't
believe that any commerically released recordings contain an actual DC offest, nor to I believe that any competent
audio chain passes true DC content.
You aren't "downsampling" here. You are downcoverting 24 bit to 16 bit. In this case, you would want to consider dithering between the steps.
Before I start, I must apologize for using the term resample, I clearly meant up/down-convert the bit depth. There is no change in sample rate being discussed here.
First of all on the subject of commercially released CDs, you would be surprised at how many releases have a measurable DC offset on their tracks - entire tracks, not portions thereof. I don't know who is mastering this stuff or encoding the content onto CDs, but they are leaving a CD offset on the music. I will not call out such CDs by name, but needless to say that pop and electronic music are ripe with DC offsets. I will admit it is not high, perhaps within 0.25%, sometimes within 0.10%. However, as part of my workflow I remove it. Perhaps an offset this low makes no difference and no audible clicks will be heard at the start/end of the track due to DC offset, but I zero it out in any case.
Next, you should be far less surprised by the fact that many recordings these days have peak amplitudes on samples that result in clipping. I choose to lower the volume on these tracks so that there are no clipped samples. Certainly, no one will argue that (perhaps many) clipped samples may be audible, depending on how a particular DAC handles them as part of conversion to analog. Sometimes I lower the overall track volume further, especially if (perhaps due to compression) its overall loudness is obscene. It should come as no surprise that many popular recordings try to maximize the perceived volume as much as possible to sound loud, even at the expense of allowing clipping to happen. I try to undo some of this damage, within reason and opportunity. Sometimes, as I mentioned previously I try to bring tracks to a common reasonable average perceived volume.
Originally Posted by bigshot
DC offset is liable to be way down close to the noise floor of the recording. Clipping is up at the top, where 16 and 24 are identical.
I understand that clipping is up at the top, but all samples are being shifted by a decimal dB amount. There must be some error in doing this. I have always thought that the error can be minimized by working at a higher bit depth. Perhaps I am wrong.
Now on to the real question. Is there any value to performing my entire process in 24-bits, which involves multiple successive operations on the data, all of which introduce potential errors into the flow due to working with integers and rounding in the process? I don't know, that is primarily my question, does it matter or can multiple changes including possible equalization be done in 16-bits with no perceivable disadvantage to the 24-bit alternative. I am open to reasonable discussion on the topic that shows me the error of my ways.
I am an audiophile with very high quality solid state equipment and headphones, but I am not willing to bury my head in the sand when a reasonable argument is presented that shows me that some of the steps I take are of dubious value. I would just like to understand the logic and thought behind them.
Edited by SharpEars - 8/22/14 at 8:14pm