F

fwrru2fr2

Audiophyte
What hardware/software do you need to rip DVD audio to a format such that you can make a CD?

The simplier and less expensive the better, although sometimes less is much less than necessary.
 
Rob Babcock

Rob Babcock

Moderator
The actual DVD-A portion is powerfully encrypted, and making a copy is a PITA. The stereo track is "crackable," but it's a lot of work and it's illegal to do so.
 
A

av_phile

Senior Audioholic
I also want to know how. I hear there is no software yet to rip the hi-res audio_ts folder in DVD-As. Much less SACDs.

But the most I can do is rip the video_ts folder where the AC3 files in some DVD-As are located. Then use a software to convert it to wav files. Often there's only one huge AC3 file for all the soundtrack, and once converted to an even bigger wav file, you then have to extract the exact track(s) you need and discard the rest.

Or you can rip the DTS files if you have the right software for it. Then use those files to make a DTS-CD or convert it to wav files using a very expensive software from surcode.

But these are essentially the same as ripping a DVD-video. DVD-As are something else.
 
Rob Babcock

Rob Babcock

Moderator
Any decent software can copy DTS- that's pretty easy. But I again must caution that cracking a DVD-A track is not easy and it's highly illegal. Many web forums will lock a post for even discussing it, as it violates US laws against decompiling/cracking copy protection. Is it technically possible? Yes. Legal- definately not.
 
WmAx

WmAx

Audioholic Samurai
Rob Babcock said:
Any decent software can copy DTS- that's pretty easy. But I again must caution that cracking a DVD-A track is not easy and it's highly illegal. Many web forums will lock a post for even discussing it, as it violates US laws against decompiling/cracking copy protection. Is it technically possible? Yes. Legal- definately not.
Huh. I did not know that people were ripping the DVD-A stereo data. Hey, thanks for the heads up. If their is an efficient software solution, I'll considering doing this myself, just to take the hi res version and convert it to 44.1/16 for use on my portable. Because, as you know, the mix is usually better sounding on DVD-A as opposed to it's CD counterpart(intentional difference).

-Chris
 
JoeE SP9

JoeE SP9

Senior Audioholic
If you don't mind loosing whatever digital surround format is encoded and maybe(?) a slight loss of fidelity just take the analog out and feed whatever burner your using. If DPL was in the original it will still be there. :cool:
 
WmAx

WmAx

Audioholic Samurai
JoeE SP9 said:
If you don't mind loosing whatever digital surround format is encoded and maybe(?) a slight loss of fidelity just take the analog out and feed whatever burner your using. If DPL was in the original it will still be there. :cool:
Surround format is irrelevant. Their are usually 2 channel mixes on music discs as a standard.

As for copying analog stream... that is anything but efficient. Have to babysit the entire time, start/stop for each track, manually enter ID tag data when converting to a usable compressed format for my portable, etc. As for (audible!)fidelity loss from a A-->D process such as this; that depends entirely on the A-D convertors used. That is yet another variable I prefer to deal with. With the hi-res digital PCM files, I could prefilter using a high quality FFT filter, then downsample to 44.1 to produce high quality converted versions of the hi-res mixes.

-Chris
 
JoeE SP9

JoeE SP9

Senior Audioholic
All True Chris
But there is nothing the record companies can do to stop you and any protection scheme is irrelevent. :)
 
A

av_phile

Senior Audioholic
I recently tried the analog route as suggested by JoeE and it works fine for me. I used to record/compile favourite tracks from LPs and CDs into open reels then to metal tapes on 3 head machines and doing the same for hi-res DVD-A proved no more difficult. Ofcourse ripping the DVD-A digital tracks should give me better sonics and faster record times but with no software in sight for these, I don't mind going the analog route. I use my player's 192/24 DAC to output the analog into my audigy LS card and, using the Goldwave software, record it at 192/24 (I think I could also record at 192/32 even) before finally converting it to 44/16. I actually would want to preserve the 192/24 files but they eat so much disk space. Now don't ask me why not just record at the 44/16. No special logical reason. It's just a comforting thought. (Who says i have to be logical in this hobby? :D )
 
U

Unregistered

Guest
If your destination format is definitely going to be 44/16 then you are better off recording it that way. If you record at 192/24, then you will have to resample and convert bit depth.

Downsampling from a higher sample rate will simply discard samples if the original rate is an integral multiple of the destination sample rate, but 192 is not a multiple of 44.1. Strike one.

Converting bit depth from 24 to 16 requires interpolation and more information will be lost. Strike two.

In reality, you will be unlikely to hear any difference between recording at 192/24 and then downsampling/bit converting vs just recording at 44.1/16 straight away, but technically its the wrong thing to do.
 
WmAx

WmAx

Audioholic Samurai
Actually, not all downconverting is created equal. I find serious audible problems to occur when downsampling/downconverting in certain software. Godlwave, for example, only caused audible problems when attempting to use it's automated downsample/downconversion options.

Cool Edit Pro 2, I found to be excellent(transparent in ABX tests compared to the original 96/24 hi res files) when first prefiltering with a high quality FFT filter to make sure no spectra over 22kHz is left. Downsampling uisng the highest quality mode. Then using a high quality 1.2 bit triangular dither to reduce to 16bit space. BTW, Dither noise must be done when A-D sampling or when reducing a word space in software. Failing to do this results in quantization error, that can become an audible problem in 16 bit space.

As you said, it is simpler to just sample at 44.1/16 to begin with, assuming one has a card that natively samples at that rate with high quality.

-Chris

Unregistered said:
If your destination format is definitely going to be 44/16 then you are better off recording it that way. If you record at 192/24, then you will have to resample and convert bit depth.

Downsampling from a higher sample rate will simply discard samples if the original rate is an integral multiple of the destination sample rate, but 192 is not a multiple of 44.1. Strike one.

Converting bit depth from 24 to 16 requires interpolation and more information will be lost. Strike two.

In reality, you will be unlikely to hear any difference between recording at 192/24 and then downsampling/bit converting vs just recording at 44.1/16 straight away, but technically its the wrong thing to do.
 
Last edited:
A

av_phile

Senior Audioholic
Unregistered said:
If your destination format is definitely going to be 44/16 then you are better off recording it that way. If you record at 192/24, then you will have to resample and convert bit depth.

Downsampling from a higher sample rate will simply discard samples if the original rate is an integral multiple of the destination sample rate, but 192 is not a multiple of 44.1. Strike one.

Converting bit depth from 24 to 16 requires interpolation and more information will be lost. Strike two.

In reality, you will be unlikely to hear any difference between recording at 192/24 and then downsampling/bit converting vs just recording at 44.1/16 straight away, but technically its the wrong thing to do.
OK, so the comforting thought has entirely disappeared. :D Thanks. A little more research to confirm.
 
A

av_phile

Senior Audioholic
WmAx said:
Actually, not all downconverting is created equal. I find serious audible problems to occur when downsampling/downconverting in certain software. Godlwave, for example, only caused audible problems when attempting to use it's automated downsample/downconversion options.

Cool Edit Pro 2, I found to be excellent(transparent in ABX tests compared to the original 96/24 hi res files) when first prefiltering with a high quality FFT filter to make sure no spectra over 22kHz is left. Downsampling uisng the highest quality mode. Then using a high quality 1.2 bit triangular dither to reduce to 16bit space. BTW, Dither noise must be done when A-D sampling or when reducing a word space in software. Failing to do this results in quantization error, that can become an audible problem in 16 bit space.

As you said, it is simpler to just sample at 44.1/16 to begin with, assuming one has a card that natively samples at that rate with high quality.

-Chris
I also have Cool Edit Pro, though I didn't find it any more user friendly than Goldwave. But I'll defer to your more thorough knowledge on the subject and try to learn more how to use it instead of Goldwave. Thanks.
 

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