[Joe Bloggs]

[CONTENTEMBED=/t/813763/measuring-digital-audio-qualities-of-bit-perfect-playback-with-diffmaker-s-correlation-depth#post_12713376 layout=inline]Quote:[/CONTENTEMBED]

Right, so when VSTHost is recording my microphone input it's actually bypassing my microphone and recording the buffered source of... WHAT?

:rolleyes:

Let's see if you can still do it right with -300dB without using VST host. VST host recording method will grab transport 0101 back into waveform in buffering as recording logical data domain. It's not real-time capturing application. Recording from I/O directly works like recording physical data domain with latency to capture packets in real-time.

Again, you've either never used VSTHost or are guilty of deliberate misrepresentation. I use VSTHost for real-time playback and recording all the time.

 

[Joe Bloggs]

For someone using nothing but VIRTUAL CABLES to accuse someone who went to the trouble of physical digital cables and trial and error with various recording plugins (at no time of which VSTHost had access to anything but the input from the PHYSICAL DIGITAL CABLE) of making "fake recordings" is so intellectually insulting it makes me want to vomit.

 

[WindowsX]

Again, you've either never used VSTHost or are guilty of deliberate misrepresentation. I use VSTHost for real-time playback and recording all the time.

 
It seems you don't understand the concept about logical data domain and physical data domain here. Nothing more to discuss then. You're deliberately avoiding actual recording solution speaks for itself.
 
Do you know that you're telling every people who don't record from VST are doing it wrong right now? Wake up dude. Get real.
 
Regards,
Keetakawee

 

[Joe Bloggs]

No, all I'm saying is YOU are doing it wrong (with foobar, for which I don't even know what recording plugin you're using). Plenty of people on gearslutz have done bit perfect digital line recordings.

 

[WindowsX]

For someone using nothing but VIRTUAL CABLES to accuse someone who went to the trouble of physical digital cables and trial and error with various recording plugins (at no time of which VSTHost had access to anything but the input from the PHYSICAL DIGITAL CABLE) of making "fake recordings" is so intellectually insulting it makes me want to vomit.

 
Can't you use reply in one post. It's getting bothersome. And I already stated to do in pure software environment. If you want to make your own rules, start a new thread propose a better approach that people can accept. I already told you before your approach is like extracting data from buffered pool not real data capturing.
 
There's nothing more to discuss with your lack of understanding the difference between these two. I can make exact data of any audio interface using VST host because data are buffered in there and waiting for me to take it back. Get it?
 
Regards,
Keetakawee

 

[WindowsX]

No, all I'm saying is YOU are doing it wrong (with foobar, for which I don't even know what recording plugin you're using). Plenty of people on gearslutz have done bit perfect digital line recordings.

 
It's not real-time bit-perfect recording. It's more like buffering data into pool and use logical capturing method instead of physical. I told you that many times before.
 
Regards,
Keetakawee

 

[Joe Bloggs]

BTW, it still makes ZERO sense that your program should be doing anything at all in the context of the software environment you're putting it in. Your sound recording processing is only buffering digital bits from one end to write them back down on the other end. Any jitter will be ignored in this process--it certainly isn't interpolating between the bits in any asychronous manner, as a cursory look at the waveforms will immediately reveal. The only reason ANY differences can exist are

1. Some processing (e.g. levelling, dither) are being carried out in the middle and cannot be turned off
2. Bits are being dropped

If (2), you wouldn't be demonstrating some obscure DiffMaker results, you would be showing those dropouts directly. Regarding (1), said dither operations are underterministic and would result in different DiffMaker results each time (as your own results show). The most plausible results here is you're running more than the trials shown and cherry-picking among the random results to put your Fidelizer in the better light.

If I weren't afraid of messing up my computer in a way System Restore won't be able to undo, I would be installing Fidelizer right now to test this.

 

[Joe Bloggs]

No, all I'm saying is YOU are doing it wrong (with foobar, for which I don't even know what recording plugin you're using). Plenty of people on gearslutz have done bit perfect digital line recordings.

It's not real-time bit-perfect recording. It's more like buffering data into pool and use logical capturing method instead of physical. I told you that many times before.

Regards,
Keetakawee

You speak as if there is any form of digital loopback recording that can do without a buffer.

Stop throwing up fake technobabble like it's real.

 

[WindowsX]

BTW, it still makes ZERO sense that your program should be doing anything at all in the context of the software environment you're putting it in. Your sound recording processing is only buffering digital bits from one end to write them back down on the other end. Any jitter will be ignored in this process--it certainly isn't interpolating between the bits in any asychronous manner, as a cursory look at the waveforms will immediately reveal. The only reason ANY differences can exist are

1. Some processing (e.g. levelling, dither) are being carried out in the middle and cannot be turned off
2. Bits are being dropped

If (2), you wouldn't be demonstrating some obscure DiffMaker results, you would be showing those dropouts directly. Regarding (1), said dither operations are underterministic and would result in different DiffMaker results each time (as your own results show). The most plausible results here is you're running more than the trials shown and cherry-picking among the random results to put your Fidelizer in the better light.

If I weren't afraid of messing up my computer in a way System Restore won't be able to undo, I would be installing Fidelizer right now to test this.

 
Why don't you try recording by yourself without using VST trick and see how things really work? There's foobar2000 and Audacity ready for you to do right now. It'd save a lot of time and trouble explaining and see the result yourself.
 
Regards,
Keetakawee

 

[Joe Bloggs]

BTW, it still makes ZERO sense that your program should be doing anything at all in the context of the software environment you're putting it in. Your sound recording processing is only buffering digital bits from one end to write them back down on the other end. Any jitter will be ignored in this process--it certainly isn't interpolating between the bits in any asychronous manner, as a cursory look at the waveforms will immediately reveal. The only reason ANY differences can exist are


1. Some processing (e.g. levelling, dither) are being carried out in the middle and cannot be turned off

2. Bits are being dropped


If (2), you wouldn't be demonstrating some obscure DiffMaker results, you would be showing those dropouts directly. Regarding (1), said dither operations are underterministic and would result in different DiffMaker results each time (as your own results show). The most plausible results here is you're running more than the trials shown and cherry-picking among the random results to put your Fidelizer in the better light.


If I weren't afraid of messing up my computer in a way System Restore won't be able to undo, I would be installing Fidelizer right now to test this.

Why don't you try recording by yourself without using VST trick and see how things really work? There's foobar2000 and Audacity ready for you to do right now. It'd save a lot of time and trouble explaining and see the result yourself.

Regards,
Keetakawee

I've been there, done that. I have seen the altered output in Audacity recording. Unlike you, I have also gotten down to the root of the problem, which is digital levelling and dither that can't be turned off.

 

[WindowsX]

I've been there, done that. I have seen the altered output in Audacity recording. Unlike you, I have also gotten down to the root of the problem, which is digital levelling and dither that can't be turned off.

 
If you've done that before, you should have known it's not possible to get -300 dB with conventional real-time capturing method. Real-time audio applications won't work like how you extract data from VST host. Even Async USB DAC can't do as much.
 
The proper methodology of measuring digital difference is how we should replicate the capturing method in the same manner as playing back as much as possible and find why things go wrong and how to improve it. You should realize that you're using irregular method that is irrelevant to the test itself right now.
 
Regards,
Keetakawee

 

[Joe Bloggs]

I've been there, done that. I have seen the altered output in Audacity recording. Unlike you, I have also gotten down to the root of the problem, which is digital levelling and dither that can't be turned off.

If you've done that before, you should have known it's not possible to get -300 dB with conventional real-time capturing method. Real-time audio applications won't work like how you extract data from VST host. Even Async USB DAC can't do as much.

The proper methodology of measuring digital difference is how we should replicate the capturing method in the same manner as playing back as much as possible and find why things go wrong and how to improve it. You should realize that you're using irregular method that is irrelevant to the test itself right now.

Regards,
Keetakawee

You speak as though VB-Audio cable isn't buffering the data before sending it to the output cable.

Any digital audio application requires a buffer.

Here, read the manual and save yourself from (feigned?) ignorance. Just search for the word "buffer".
http://vb-audio.pagesperso-orange.fr/Cable/VBCABLE_SystemSettings.pdf

 

[WindowsX]

You speak as though VB-Audio cable isn't buffering the data before sending it to the output cable.

Any digital audio application requires a buffer.

Here, read the manual and save yourself from (feigned?) ignorance. Just search for the word "buffer".
http://vb-audio.pagesperso-orange.fr/Cable/VBCABLE_SystemSettings.pdf

 
I used different wording between 'extraction' and 'capturing'. I also explained before that using real-time recording will have latency comes into play. I hope that's all I need to to tell you. Your method is different from how most real-time audio playback/recording applications work. Don't you agree?
 
Anyway, I've gotta go to continue my job now. Happy listening.
 
Regards,
Keetakawee

 

[Joe Bloggs]

So when I use VSTHost to record and process my microphone input in real time to improve my Skype call quality in real time, I'm not doing real time audio? I can probably get the latency down lower than any method you're using! Heck I play movies and games with sound processed by VSTHost too, do you think I can live without realtime audio in those circumstances?

 

[WindowsX]

So when I use VSTHost to record and process my microphone input in real time to improve my Skype call quality in real time, I'm not doing real time audio? I can probably get the latency down lower than any method you're using! Heck I play movies and games with sound processed by VSTHost too, do you think I can live without realtime audio in those circumstances?

 
That's totally different topic. There's nothing wrong in adding VST into audio chain for playback/recording. It'll change from just 'audio stream' to 'audio stream + dsp in vst host' before streaming. But recording audio from vst host software is different. Your method was synchronization within device. All you got was aligned data with the same clock synchronization within the same device. You might as well just copy/paste that file instead of doing playback/recording simulation.
 
There's no record of anyone successfully doing bit-perfect recording with -300dB across the device no matter how they configured it. You can try playing audio with foobar from machine A using audio interface with digital output and perform recording from machine B using audio interface with digital input. In this scenario, I'll allow you to use VST or whatever you want if you can make bit-perfect -300dB recording using different machines and host controllers.
 
Regards,
Keetakawee