ASIO Music Player Plugin (1 Viewer)

[Telstar]

Windows ate my reply...


Bah, anyway, look also here (best link so far):
http://www.audioasylum.com/cgi/vt.mpl?f=pcaudio&m=23886

 

[grubi]

Can anybody tell me in short what sould be the point in upsampling?
I would consider best to always have no change in sampling rate from source material at all.

grubi.

 

[Telstar]

It can create the illusion of a more natural sound to the listener. It is very subjective. There are advocates of NOS (non-oversampling/upsampling) too.

I dont know in which side i am because I haven't listened to an OS/US source.

Anyway, to notice something, you would need a high end system.

 

[floepie]

Apparently, we have a deciphering threshold of 22 bits level per sample. And, with an interpolator that's "smart" enough and can more precisely match the new sample values with the original wave, one would think that on paper at least, one could hear a difference in sound quality, especially at high frequencies, as a 44.1 kHz sampling rate doesn't exactly approximate the analog electrical wave that well. But, as the above poster stated, I'm not sure what kind of audio equipment you would need in order to hear a real difference.

 

[Symphy]

I won't claim to be an expert in these kind of things, but this is the way i see things:

If any upsampling is done at all it should be a multiple of the original samplingrate. I believe upsampling from 44100 to 48000 will only destroy things. However, when upsampling from 44100 to 88200 the original samples will remain and only new samples will be created inbetween. This kind of upsampling is also referred to as oversampling

...especially at high frequencies, as a 44.1 kHz sampling rate doesn't exactly approximate the analog electrical wave that well...

Oversampling can never restore any detail that is lost at the ADC and therefor not in the data, that's physically inpossible. It can however help recreate the waveform in the analog domain in a more accurate way. And thats why many DACs have their own built-in oversampling logic (2 or 4 times oversampling).
Now if a DAC is designed to handle 44100 at a certain level of quality, it will either do its own oversampling or use some other method to achieve the same result. Feeding it with our own created 88200 will most likely cause the DAC to kick down its own oversampling logic, and the result will be exactly the same. And if the DAC's oversampling algorithm is better then our software algorithm, the result will actually be worse...
Now i don't know what the effect is on single-bit DAC's and digital filters etc. but whether it is usefull or not will probably depend completely on the particular DAC used.

Doing software upsampling may be very usefull though for ppl building their own DACs. A non-oversampling DAC without analog filtering is not that difficult to build. Combining that with a good software oversampling will probably lead to very good results.

The upcoming PureAudio release will have an option for 2 times oversampling (in floating-point so at 24 bit resolution!) , so anyone can do the test and decide for themselfes
First however i have to figure out what goes wrong when feeding floating-point data to ASIO, something is not right there...

Regards,
Symph

 

[Telstar]

I won't claim to be an expert in these kind of things, but this is the way i see things:

If any upsampling is done at all it should be a multiple of the original samplingrate. I believe upsampling from 44100 to 48000 will only destroy things. However, when upsampling from 44100 to 88200 the original samples will remain and only new samples will be created inbetween. This kind of upsampling is also referred to as oversampling

True and I agree. About bitrate from 16 to 24 is x1.5 and that is also claimed to produce results (from OS advocates). This is referred as upsampling.

Oversampling can never restore any detail that is lost at the ADC and therefor not in the data, that's physically inpossible. It can however help recreate the waveform in the analog domain in a more accurate way. And thats why many DACs have their own built-in oversampling logic (2 or 4 times oversampling).

Also true.

Now if a DAC is designed to handle 44100 at a certain level of quality, it will either do its own oversampling or use some other method to achieve the same result. Feeding it with our own created 88200 will most likely cause the DAC to kick down its own oversampling logic, and the result will be exactly the same. And if the DAC's oversampling algorithm is better then our software algorithm, the result will actually be worse...

Yes, that is why an oversampled signal should only be sent to DACs that accept higher frequency sources, otherwise the result is detrimental.
Software OS/US can be useful to save the money to buy an OS DAC, but the key is that the used DAC must accept the higher resolution/freq signal. The funny thing is that I havent seen any non-OS dac that accept those signals

 

[grubi]

Symphy:

Thank's for making things clear again. This exactly describes my point of view.
Conclusion for me: On high end equipment keep original sampling rates and try
to transmit everything unaltered.

Regards,
grubi.

 

[Symphy]

Hi!
I have a problem, and is not sound related.
When i enable the plugin, i go to my Music and pick the first song in a folder, it just scroll down to the botton and plays nothing.
I try diff skins but nothing, do you think could be svn related?

Cheers!

Sorry for the late reply...
Could it be you have selected ASIO while no ASIO device is present on your system? If not, pls post some debuglevel-logfiles. It's certainly not related to skins. Version 0.1.2.0 requires MP 0.2.3.0 RC3

Symphy

 

[felo]

Hi!
Thnks for answer!
Audio card: Audigy 2 ZS
Skins i try: BlueTwo & Xface.
MP version: 0.2.3.0 Final (no SVN)
Windows Ver: Windows XP SP2 all Updates.

Maybe my configuration is wrong, ill try dif confs.
Thnks!

 

[Symphy]

Audio card: Audigy 2 ZS

In case you have selected "Use ASIO device" in the plugin config, try the following: go to Advanced setting/ASIO and set the minimum supported samplingrate to 48000.

As far as i know the Audigy ASIO drivers do not support 44100 and this way the plugin will upsample to 48000. Note however that using ASIO does not have much benefits this way, you might as well select "Use windows device".

The logs you provided do not contain any data of PureAudio plugin. To record it:
Enable the plugin, start Mediaportal and let it go wrong in My Music, then close MediaPortal again and make a copy of the logfiles. They get overwritten with fresh ones each time you start MediaPortal.

Regards,
Symphy