Use of 88.2 kHz sample rate

[Cary B. Cornett]

I am a new user of DC 10.8 (I had version 6 some time ago).
I am doing a flat capture of an LP at 88.2k and trying to process at that rate. None of the automated click removal processes, even with backwards processing, seem to do much. I have tried various settings, and the only mode/setting that seemed to get the clicks also distorted the audio. I did do one manual click removal, which did work. Doing the EQ processing with the VPP seems to work fine. I am wondering if DC is picky about the sample rate used.
Does anyone here process using the 88.2k sample rate?
Has anyone here tried using this rate and found it did not work at that rate?

[Craig Maier]

Use the the 96 kHz sampling rate. Some sound-cards do not support 88.2 kHz. When you process using the EZ Impulse filter at 96 kHz and have completed that task, the file is easily converted to 44.1 kHz.thereafter. The change sample rate/resolution feature is found under the Edit menu.

Craig

[Craig Maier]

Some soundcards do not work on 88.2 kHz, but they will work at 96 kHz. After the use of the EZ Impulse Filter, you can down convert the file to 44.1 kHz.

Craig

[Craig Maier]

This computer that I am using employs a Realtek chipset for sound. I just tried and successfully recorded at 88.2 kHz and used (with very good results) the EZ impulse filter on that file. So, I am wondering what brand your sound chipset or soundcard is? Can you get back to us on that?

Craig

[Cary B. Cornett]

In answer to Craig:
My sound care is a MOTU 424 with a 2108 MKIII interface. The ASIO drivers for this work transparently at 44.1, 48, 88.2, and 96 kHz. I recently learned the hard way that WDM driver support is much less versatile. Naturally DC10 doesn't "speak" ASIO. So, I do my capture with SawStudio instead, and then can load the resulting 88.2k file into DC10. My playback workaround is to use Blue Cat Patchwork, which DOES speak ASIO, and a plugin utility called WDM2VST. This means DC10 sees a WDM driver, and I can get my playback at whichever sample rate I choose.

Playback, then, while a bit cumbersome, is not a problem. Why am I being stubborn about the sample rate? Because the math for sample rate conversion is less, um, fraught at integer conversion ratios, in this case exactly 2:1. (I have been called "pedantic" for thinking this way.)

Since starting this thread, I have spent a little more quality time with DC10 and its documentation. While there are some really neat things about this program, I am finding it somewhat finicky and fussy in some ways. It sometimes crashes and shuts itself down, and today it froze and cost me a couple of hours of work (I wan't smart enough about doing periodic file saves and trusted the express mode too much). To use DC10's advantages, I'll just have to accept its quirks.

I still need an answer to my original questions, rather than advice to just "go 96k". I have found that I can run the EZ impulse filter, but I cannot audition settings live. In order to get results I can hear, I have to just make settings and process the file. I can get its automated process to eliminate some clicks and reduce others, but I am also seeing thousands of click detections in the first 30 seconds or so of a file (the beginning of side 1 of The Yes Album), which suggests to me that some damage is being done to the music. I am doing the click processing to a flat transfer, and when I'm done with all the impulse stuff, THEN I plan to use the VPP to process the EQ.

What seems to work best so far is the manual "search and destroy" process for clicks. Find the click, zoom in really tight, highlight the offending part, and hit the O key. When I select the boundaries correctly, the repair is absolutely undetectable to the ear! Now if only I can get the automated detection tools to work as well. It might be easier if I could live audition the settings, which takes me back to "can this be done at 88.2k?"

I still need to hear from anyone who has worked with DC10 at this sample rate.

[Cary B. Cornett]

I can record at 88.2k using a different program.

[Craig Maier]

Pedantic is a good word for this. The conversion from 96 kHz down to 44.1 performed with 32 bit, dual precision math means that the rounding error will be 384 dB down from a 0 dB reference. If you can hear that error, well, you are a very unique individual. Personally, I have measured my own sensitivity to such errors and I can not detect anything lower than 75 dB down from the 0 dB reference.

Craig

ps - 88.2 kHz works fine on my Hp laptop with a realtek chipset; try using wdm drivers along with your computers native audio chipset..

[Marc Hildebrant]

In my restoration method, I use 96 kHz sampling with good results. Various filters and methods are shown in my book, available on this website.

Marc

[Marc Hildebrant]

I show some examples of 96 kHz sampling in my book, available on this website.

Marc

[Cary B. Cornett]

Maybe my concern about sample rate conversions is a bit, er, hidebound, but I remember when loss of quality in sample rate conversion was a real concern among recording engineers. Ever after the first discussion I had on that topic, I have been cautious about such conversions, particularly between sample rates that have no simple factor relationship between them. Then again, back then one was lucky to get 14 bit accuracy out of 16 bit converters. The tools now are indeed much better.

[Craig Maier]

The error using the Diamond Cut File sample rate converter is 384 dB down from a 0 dB reference when going from 96 kHz down to 44.1 kHz. The finest instruments on the planet could not measure that degree of quantization error level. Besides all that, a brand new vinyl LP on a very good day right out of the sleeve will yield noise levels on the order of only -55 dB. So, it is an irrelevant point.

If a power signal could be produced 384 dB above the noise floor reference of a sound source having the power produced by the hind-legs of a flea, it would destroy the planet earth due to the energy contained therein (think extremely large thermo-nuclear H-bomb)..