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Author Topic: Problems upsampling CD Audio with MJ Plus 8.0.329  (Read 1152 times)

Osho

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Problems upsampling CD Audio with MJ Plus 8.0.329
« on: July 20, 2002, 04:15:18 pm »

Hi,

I have MJ Plus version 8.0.329 running on Windows 2000. I have a Lynx L22 sound card, which supports rate upto 24-bit/200kHz. I am using this sound card's analog output for CD Playback. CD Audio playback with default settings works great. I tried to upsample CD Audio by changing  Settings -> Options -> Playback -> Bitdepth to 24 bit. This result in Very high spikes in sound output which keep on increasing in volume. I can not hear any music. It almost sounds like it will definitely damage the system. The Output Mode is Wave Out (default). Any ideas what is going on here?

thanks,
Osho
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Bill Ko

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #1 on: July 20, 2002, 08:18:15 pm »

I'm not sure why you would want to do this because 16 bit depth is maximum depth for an audio CD.  Anything more would not make it any more accurate, and might cause sampling problems.  (A 32 bit conversion might work better.)

My advice would be to stick to default bit-rates.  They always work best, unless something happens where the default rate is incorrect.

Bill
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Osho

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #2 on: July 20, 2002, 11:56:48 pm »

I wanted to do this just to get an idea what upsampling does to the sound quality. I am still curious. I couldn't upsample to 32 bit as my sound card supports only 24 bit.

Also, is there a setting somewhere where I can upsample the frequency - say from 44.1 to 88.2 or 166.4 ?

thanks,
Osho
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Bill Ko

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #3 on: July 21, 2002, 06:18:32 pm »

Okay, this is what you want to here (hopefully).

The bit depth refers to the value range each sample can be.  For 16 bits, the samples range from 0 to 64k, or 2^16.  Since audio is based on the sine wave, values would more accurately be described as ranging from -32k to |PLS|32k.  The values represent the given value of the wave form for that sample - the "loudness" of that sample.

The sampling frequency refers to how many samples per second (expressed in Hertz, Hz).  Convention says that the sampling frequency must be at least twice the highest frequency of the sample to represent the waveform accurately.

All this being said, I'll give you an analogy of what you are attempting to do.  What you are doing is analogous to converting a GIF picture (256 colors) to a JPG (millions of colors) picture.  You're only going to get 256 colors out of the millions because the original picture only had 256 colors.  It's not a perfect analogy, but it gets the point across.  You can't get BETTER quality than the original sample.  You just can't.

Changing the sampling rate of a digital waveform would do one of two things, depending on the criteria.  Reading the same amount of samples in a shorter time would transpose the song into a higher "key" (although it's not strictly true, because the increment may not be an actual music key), in a proportionately shorter time - like playing a 33rpm record at 45rpm.  If you tried cramming more samples in the same time frame, you end up with the analogy mentioned in the above paragraph.

So this is the bottom line: digital waveform re-sampling from lower to higher bit rate = waste of time.  (Higher to lower has its advantages, though.  You save lots of disk space - with the tradeoff that the resulting waveform is described less accurately than before - but you might not be able to notice the difference.)  As for changing the sampling rate, I *did* find one use for it.  Some of my (former) band's stuff was mastered with uncalibrated equipment (if I could find the guy, I'd kill him!), and the speed, when played back on a cassette player (it was that long ago, okay!), it sounded like it was playing at the wrong speed - way too fast!  So I digitized it, then changed the sampling rate to "stretch it back out" to what it should have been (relying upon my ears to tell me when the music was being played back at the correct pitch - not very scientific, huh?)

Bill

Oh, I forgot to mention - for all A to D conversions (analog to digital), use the highest sampling rate and bit depth you can handle for the highest quality.  Just remember that the higher the quality, the more disk space you consume - and, if you sample above 16 bits or 44kHz, you will have compatibility problems with sound cards that cannot process those files.  Unless you use MJ!  (I actually haven't tried that yet, though!)
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JimH

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #4 on: July 21, 2002, 06:34:03 pm »

Bill,
Wow!  What do you do for a living?  Sound engineer?

Thanks for the details.

Jim
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Jim Hillegass
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Osho

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #5 on: July 22, 2002, 10:14:36 am »

Bill,

What you are referring to is converting a lower sampled signal to a higher sampled/bit-depth one without adding anything new to the data. In that case, your analogy of GIF to JPEG is appropriate.

However, good algorithm indeed takes the advantages of more sample points/bit resolutions available and in fact adds it's own data "intelligenetly" using interpolation. Of course, the sound quality generated is very software dependent and how it does this. Many sound cards also do that in their final stages of DAC.

Some high end CD Players also do this to achieve more of an "analog" sound. For example, MOON Eclipse CD player converts a 16-bit/44.1kHz CD Audio signal to 24-bit/352.8kHz (8 times 44.1). There is a white paper (PDF) at http://www.simaudio.com/pdf/Upsampling.pdf showing the benefits of doing such a conversion for *CD AUdio Playback*

I wish that I can use MJ to convert from 16-bit/44.1kHz to at least 24- bit/166.4kHz (200kHz is the max that Lynx L22 soundcard supports).

Osho
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Matt

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #6 on: July 22, 2002, 11:12:47 am »

MJ has a pure 32-bit playback chain, and it down converts to whatever bit depth it needs on the way out.  With DSP's disabled, the transition from 16 bit (the CD) -> 32 bit (internal) -> 16 bit (output) is bit perfect.  However, with DSP's enabled, it's better to downscale the output as little as possible, so there isn't quantization noise.  So, higher bit depth output has some merit.

Osho, try DirectSound to see if it works any better at 24-bit.  Also, check for newer drivers for your Lynx.

About over-sampling / resampling, that's probably best left to your DAC instead of Media Jukebox.

Take care.

-Matt

(p.s. thanks for that great post Bill)
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Matt Ashland, JRiver Media Center

Osho

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RE:Problems upsampling CD Audio with MJ Plus 8.0.329
« Reply #7 on: July 22, 2002, 01:03:34 pm »

Thanks Matt for the Tip on DirectSound. I will try that today.

Could you please explain what do you mean by "MJ has a pure 32-bit playback chain" ??

Osho
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