[frankR]

I finally got my Audiotrak HD2. I'm very pleased with the sound, just using the headphone out, but would like to aquire a deticated head phone amp someday and upgrade the opamps on the HD2. I have a EMU 0404 usb on back-order from Muscian's Friend that I may cancel now.

So I'm wondering where I can find a good software interpolator that will upsample all my audio to 192khz. I found a program called foobar that can upsample to 96khz, but how about the full 192khz? I've tried the ASIO plugin with foobar and winamp, but I can't make it work. I think it installed correctly, it just isn't working, sometimes I get no sound at all, maybe thinking I don't have it configured correctly?

 

[sejarzo]

The general opinion seems to be that while most of the current generation chips can handle 192 kHz sample rates, their performance is not as good as it is at 96 kHz.

The Benchmark DAC1 runs an asynchronous sample rate converter to operate at 110 kHz, where the Benchmark engineers found was the optimal point.

AFAIK, none of the Foobar upsampling plug-ins for version 0.8.3 will go beyond 96 kHz. I can't get any 0.9.x versions to work well with my 0404 USB, so I've not investigated all of the latest options in upsampling plug-ins.

 

[Bones13]

Do some creative searching for SRC (secret rabbit code - no kidding even) if you can still find it. This program takes a fair amount of processing power though. I tried it a while back, there are several discussions of using here on the board. Many swear by upsampling if your DAC will take it. I was gaming a good bit back there, and EQ needed too much of my processor to run SRC, so I stuck with bit perfect SPDIF.

 

[frankR]

I found the secret rabbit code, "code" literally. It comes uncompilled. I don't have a C++ compiler on my machine. It would be nice if there was compiled version out there.

 

[frankR]

The block diagram from the AKM AK4396 shows an 8x interpolator before a delta-sigma modulator. It looks like the DAC upsamples internally. I wonder how much of difference upsampling in software acutally makes?

AK4396

I can hear the difference between 48khz and 96khz (upsampled), espeacially on a high frequencies, like a ride cymble, it sounds smooth less metallic, more lifelike. I'd be interested to hear 192khz or even 110khz.

 

[LordofDoom]

So... wait, is 96000hz upsampling better sound quality than 44100hz when using Foobar2000 Upsampler (Rabbit)?

 

[sejarzo]

frankR, do you really mean 48 kHz? Material ripped from RedBook CD's is native 44.1k, and resampling it to 48k usually produces a bad result.

That's why I think if there is any benefit, upsampling to 88.1k or 176.2k makes much more sense than 96/192. The interpolation of data between the original data points is a lot simpler than doing sample rate conversion along with upsampling.

I've also wondered about that term "interpolator" on the AK4396 diagram--it sure makes it seem as if it is producing additional samples at different values between the original data points. However, I was under the impression that those functions within DAC chips typically only increased the data rate without calculating any intermediate points, so that filtering out the sampling noise was easier (noise at a much higher frequency than the audio range, so a gentler filter with a higher cutoff could be used.) Anyone out there know any better?

 

[b0dhi]

If you look at the output of the AKM4396 on an oscilloscope when feeding it, say, a 22Khz input, you'll see a smooth 22Khz sine wave at its outputs. However, when you feed it, say, a 2Khz square wave as input, at the outputs you'll see that the chip has smoothed out the corners of the square wave (in 44.1Khz mode, that is). The SRC or Shibach upsamplers, when running in high quality 192Khz mode, will get the AKM to still produce a smooth sine wave at 22Khz but also allow it to produce a more square-like output when necessary, due to the higher bandwidth. Even though it uses half my CPU, I use the upsampler in 192Khz high/ultra quality mode when doing serious listening.

 

[frankR]

Quote:

Originally Posted by LordofDoom /img/forum/go_quote.gif So... wait, is 96000hz upsampling better sound quality than 44100hz when using Foobar2000 Upsampler (Rabbit)?

I think it sounds better.

 

[frankR]

Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif frankR, do you really mean 48 kHz? Material ripped from RedBook CD's is native 44.1k, and resampling it to 48k usually produces a bad result.

I look at as resampling a video signal. Depending on the alogorithm, it could produce a good result or a bad one.

 

[frankR]

Quote:

Originally Posted by b0dhi /img/forum/go_quote.gif Even though it uses half my CPU, I use the upsampler in 192Khz high/ultra quality mode when doing serious listening.

Would you please tell me how you're doing this?

 

[b0dhi]

Quote:

Originally Posted by frankR /img/forum/go_quote.gif Would you please tell me how you're doing this?

I'm using Winamp, so I don't know if it will apply to you, but I do remember that SRC (or was it SSRC?) in Foobar could also do 192Khz (except it would crash when playing FLACs, even if I tried older versions of Foobar). The output plugin I use is here: WINAMP5用 ASIO出力プラグイン (dll version) - ãŠãŸã¡ã‚ƒã‚“ã®MIDI/Audioソフト. It's a combined ASIO output and high quality resampler.

 

[frankR]

Quote:

Originally Posted by b0dhi /img/forum/go_quote.gif I'm using Winamp, so I don't know if it will apply to you, but I do remember that SRC (or was it SSRC?) in Foobar could also do 192Khz (except it would crash when playing FLACs, even if I tried older versions of Foobar). The output plugin I use is here: WINAMP5用 ASIO出力プラグイン (dll version) - ãŠãŸã¡ã‚ƒã‚“ã®MIDI/Audioソフト. It's a combined ASIO output and high quality resampler.

Thanks b0dhi, that's exactly what I was looking for. I actually found that link the other night, but I couldn't read it so I never downloaded anything!

I'm running at 192khz, it isn't very CPU intensive, I'm hearing a couple of pops every once in awhile. I may go back to the foobar converter.

Edit: I switched to Ultra quality, but it played jumpy. I increased the buffer size to 63 and changed the thread priority, that seems to have helped, still hearing pops, like bit drop out.

 

[b0dhi]

To fix that you need to increase the latency on your sound card. If you're using ASIO4All, it's the "buffer size" setting. Otherwise, there should be a setting in your sound card's control panel. I set mine to 2048 samples and it doesn't skip at all.

 

[frankR]

The maximum sample size availible is 1024. I played with the sample size. It still was popping though. The Foobar upsampler seems be a lot more stable for me ATM. I'll try the Winamp plugin.

Do I have to have ASIO4ALL installed. I think I uninstalled it the other night because I had no idea what I was doing.