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To anyone actually knowledgeable with this: Is there any equalizer for DSD format that does not...

post #1 of 10
Thread Starter 

To anyone actually knowledgeable with this: Is there any equalizer for DSD format that does not use PCM conversion while operating?

 

More I know about DSD, less supportable I become toward this format.

 

It seems there isn't many (if any) audio equipments that can take care of DSD in native stream, such as noise shaping, equalizing, etc.

I mean what's the point of having DSD when the file itself has to be edited by convert it to PCM and then convert back to DSD? As far as I know, PCM -> DSD is NOT lossless conversion.

I believe people buying those DSD format is getting shorter end of the stick. 


Edited by wnmnkh - 3/22/14 at 4:52pm
post #2 of 10

You have summed up the situation well.

post #3 of 10

If human ears can't hear the difference between DSD and PCM, then I don't see what difference it makes. There's no practical advantage to making a fetish of file size. The thresholds of human perception set the line that divides what matters from what doesn't, not the file format. Sometimes I get the impression that audiophiles who know a lot about esoteric high end digital technology don't know much at all about their own ears.

post #4 of 10
Thread Starter 
Quote:
Originally Posted by bigshot View Post
 

If human ears can't hear the difference between DSD and PCM, then I don't see what difference it makes. There's no practical advantage to making a fetish of file size. The thresholds of human perception set the line that divides what matters from what doesn't, not the file format. Sometimes I get the impression that audiophiles who know a lot about esoteric high end digital technology don't know much at all about their own ears.

 

I understand your point. But the fact that we are not getting real DSD (PCM converted into DSD) is.. this is downright fraud. Regardless of we can hear the difference or not, if the source is not trustworthy, there is literally no reason to invest in high-res audio.

 

Personally I fully understand CD quality is already overkill in many situations, and anything higher would not make any difference. But I do want to get the best quality possible if I like the music, and I believe there are people having same thought as me. If that quality is faked, I mean... this is not just a matter of scientific discussion anymore.

post #5 of 10
You're talking about native DSD recordings, right? I'll bet they try to sell upsampled DSD next. High bitrate audio is a joke. Just take a good mastering and knock it down to a 320 LAME or AAC file and listen the hell out of it. The music is what matters, not the digital bubble wrap they package it in.
post #6 of 10
Thread Starter 
Quote:
Originally Posted by bigshot View Post

You're talking about native DSD recordings, right? I'll bet they try to sell upsampled DSD next. High bitrate audio is a joke. Just take a good mastering and knock it down to a 320 LAME or AAC file and listen the hell out of it. The music is what matters, not the digital bubble wrap they package it in.

 

They are already selling 'upsampled' DSD, since without it, it seems there is no way to remove upper freq noise caused by DSD algorithm itself. I see like 2x, 4x, 6x (yes, 6 bits) DSD selling with more than 30 bucks.

 

Well, 320 is fine... but I just do not like the fact that we cannot get a source file that does not degrade after transcoding.

post #7 of 10
Have you ever done a test on transcoding? I have... I took a very good classical orchestral recording and I encoded it to AAC 256 VBR then to AIFF then back to AAC 256 VBR ten times. Tenth generation. You know what it sounded like? Almost identical to the first generation rip.

I don't think anyone else has done any tests on transcoding. They just rip to the max and assume it's better. For most purposes, there really isn't a point to lossless rips. Especially if you own the CD you are ripping.
post #8 of 10
Quote:
Originally Posted by bigshot View Post

Have you ever done a test on transcoding? I have... I took a very good classical orchestral recording and I encoded it to AAC 256 VBR then to AIFF then back to AAC 256 VBR ten times. Tenth generation. You know what it sounded like? Almost identical to the first generation rip. ...

 

You used the same encoder each time. After the first run, everything that could be left out had already been left out.You need to use a different encoder or set of parameters for each generation. The idea is that each encoder has a different set of filters and models to decide what to leave out. It can be a problem in broadcast distribution chains where multiple encoders with different parameters may be cascaded.

post #9 of 10

Ah, that's interesting. It isn't generation loss, it's a one time loss caused by the codec. But that means if you are ripping your CD to a codec that is audibly transparent, like AAC 256 or LAME 320, then it would down transcode to other codecs and bit rates pretty much the same as a lossless rip would. The only quality degradation would be in transcoding to low bit rate lossy formats with audible artifacting. I never hear anyone say that when it comes to lossy vs lossless arguments. They always say you have to rip to lossless because then you can transcode to other formats without generation loss.

post #10 of 10
Quote:
Originally Posted by bigshot View Post
 

Ah, that's interesting. It isn't generation loss, it's a one time loss caused by the codec. But that means if you are ripping your CD to a codec that is audibly transparent, like AAC 256 or LAME 320, then it would down transcode to other codecs and bit rates pretty much the same as a lossless rip would. The only quality degradation would be in transcoding to low bit rate lossy formats with audible artifacting. I never hear anyone say that when it comes to lossy vs lossless arguments. They always say you have to rip to lossless because then you can transcode to other formats without generation loss.


Unfortunately, transcoding lower and lower doesn't quite work the way you would hope. Each encoder assumes the input hasn't been through an encoder already.

You could try it:

Try 1: Encode to LAME 128.

Try 2: Encode to AAC 256 then encode the result to LAME 128.

I suspect you'll find the two sound somewhat different.

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