[UncleFestive]

Hello all, I am a chronic lurker around here, usually don't have much to add, but I had to pass this on, maybe some of you who I have learned from over the years will find value in it!

I have a Gateway Laptop with an Audigy 2 ZS PCMCIA sound card. I've always used the analog out to my Heed Canamp for music listening. I recently started using the optical out to an old JVC receiver, with MediaMonkey using Kernel Streaming Output V 3.63, and listening to my HD650's thru the receiver's headphone jack in preparation for adding my Audio GD Compass when it arrives.

I really want Bit Perfect playback but I've always read that no matter what, a Creative sound card up-samples to 48khz. I'm going to debunk that myth, and declare my Audigy is passing Bit Perfect to my receiver. You can too by reading and following the steps on this page:

FAQ: Kernel Streaming and the REVO - AVS Forum Archive

The concept is pretty simple: If you play a DTS encoded signal and it is resampled through your soundcard or your computer (kmixer or other), you'll get nothing but white noise. If it passes through bit perfect it will play as intended. Be aware the digital signal must be input into a receiver capable of decoding DTS. Most home theater receivers are capable of this. A more in-depth explanation of ASIO and Kernel Streaming, along with links to download DTS encoded media are provided on this page.

I downloaded one of the songs and ran the test. Digital out to the receiver yielded perfect reproduction of the source! Bye Bye Creative resampling myth! Bit Perfect Rules!

Now if I only had my Compass....

 

[Szadzik]

I am using an Onkyo HTX-22HD with my computer and used AC3Filter in order to get DTS straight through to the Onkyo for it to decode the 5.1 sound, no problems with it.

 

[1stimer]

thanks OP....trying to figure out if/what I need to better my CD players sound quality (Arcam CD73), by streaming files from computer to PS3 -->Dac--->integrated using PS3 Media Server.

I can confirm that I am getting no upsampling of the digital source in the PS3 and it is delivering bit-perfect into the reciever.

Now I just need to add a dac to the mix so I can compare to my CDP.

The PS3 is a bit too noisy...wonder what I can do about that.

Thanks again.

 

[ericj]

hooked up to either my old sony or either my old or new Onkyo HT receivers, i just press the 'display' button on the receiver remote, and it tells me what it's receiving.

'course, it usually displays it for a second or two when the stream first starts, too.

The article is misleading because the old dolby 5.1 signals use the same clock as 48khz stereo pcm, so an SPDIF pathway that only supports that clock will do either 48khz s16le or dolby digital, no problem.

The 1.5mbit DTS signals use the same clock as 96khz s16le, iirc.

If your receiver tells you that it's receiving 44.1khz pcm, it's probably receiving 44.1khz pcm.

(I lurve my TX-SR707)

 

[krmathis]

Perfect reproduction here, using the MacBook Pro and Isabellina.
Playing back the "The Other Side - DTS (WAV @ 44.1kHz/16-bit Stereo)" file from Free DTS and AC-3 (Dolby Digital) Downloads - Kelly Industries

 

[grokit]

Quote:

Originally Posted by krmathis /img/forum/go_quote.gif Perfect reproduction here, using the MacBook Pro and Isabellina. Playing back the "The Other Side - DTS (WAV @ 44.1kHz/16-bit Stereo)" file from Free DTS and AC-3 (Dolby Digital) Downloads - Kelly Industries

I'm becoming convinced that computer-based music servers are much simpler to implement on a Mac. Must be why most pro recording studios use them

What I've always understood is that Redbook stereo CDs are 44.1kHz/16-bit/2-channel (stereo) audio files, and with the extra space available on DVDs they went to 48kHz/16-bit multichannel audio files for surround sound. That's why the adaptive USB audio standard supports both 44.1kHz/16-bit and 48kHz/16-bit files.

So I am probably over-simplifying but that's my 2¢

 

[.Sup]

My DacMagic upsamples so there's no way to get bit perfect output. But I don't really care, the sound is great

 

[MCC]

My X-Fi Titanium is bitperfect via ASIO and automatically switches the samplerate to match the source. That is, of course, when the abysmally terrible drivers aren't trying to screw me over.

 

[SirDrexl]

Quote:

Originally Posted by .Sup /img/forum/go_quote.gif My DacMagic upsamples so there's no way to get bit perfect output. But I don't really care, the sound is great

But, the point of the test is to find out if your source is outputting bit-perfect audio. Even if your DAC will do its own upsampling, you could use the test to make sure that it's being fed a bit-perfect signal. Otherwise you'd be resampling twice.

 

[Criswell]

I just wanted to add, that forum color layout is effing terrible! And I will be back with my tests for bit perfect when I get home from work

 

[sonci]

Quote:

Originally Posted by UncleFestive /img/forum/go_quote.gif Hello all, I am a chronic lurker around here, usually don't have much to add, but I had to pass this on, maybe some of you who I have learned from over the years will find value in it! I have a Gateway Laptop with an Audigy 2 ZS PCMCIA sound card. I've always used the analog out to my Heed Canamp for music listening. I recently started using the optical out to an old JVC receiver, with MediaMonkey using Kernel Streaming Output V 3.63, and listening to my HD650's thru the receiver's headphone jack in preparation for adding my Audio GD Compass when it arrives. I really want Bit Perfect playback but I've always read that no matter what, a Creative sound card up-samples to 48khz. I'm going to debunk that myth, and declare my Audigy is passing Bit Perfect to my receiver. You can too by reading and following the steps on this page: FAQ: Kernel Streaming and the REVO - AVS Forum Archive The concept is pretty simple: If you play a DTS encoded signal and it is resampled through your soundcard or your computer (kmixer or other), you'll get nothing but white noise. If it passes through bit perfect it will play as intended. Be aware the digital signal must be input into a receiver capable of decoding DTS. Most home theater receivers are capable of this. A more in-depth explanation of ASIO and Kernel Streaming, along with links to download DTS encoded media are provided on this page. I downloaded one of the songs and ran the test. Digital out to the receiver yielded perfect reproduction of the source! Bye Bye Creative resampling myth! Bit Perfect Rules! Now if I only had my Compass....

Try to use files of different sample rate, not only the 48Khz which is the goal of the Audigy,

PLEASE if someone could test Wasapi for bit perfect output, so I can finally change to 7,
THANKS

BTW, happy new 2010, finally 2012 is coming..

 

[SirDrexl]

Yes, that's a good point. Make sure you try the 44.1khz WAV files. If a card resamples to 48khz, it will probably pass 48khz DTS with no problem.

I get bit-perfect output with WASAPI and Windows 7. That's with a Xonar D2, using the Foobar plugin, with the device set to "S/PDIF Pass-through." It doesn't work if the device is "speakers" (the default), so I think the card is doing some kind of conversion to everything using that device.

 

[sonci]

Quote:

Originally Posted by SirDrexl /img/forum/go_quote.gif Yes, that's a good point. Make sure you try the 44.1khz WAV files. If a card resamples to 48khz, it will probably pass 48khz DTS with no problem. I get bit-perfect output with WASAPI and Windows 7. That's with a Xonar D2, using the Foobar plugin, with the device set to "S/PDIF Pass-through." It doesn't work if the device is "speakers" (the default), so I think the card is doing some kind of conversion to everything using that device.

Yes, I've noticed that with vista, only selecting spidf as default device got the automatic convertion rate in Creative audio creation mode,
at least wasapi is bit perfect after all..

 

[grokit]

Not to hijack or be disrespectful, but what's the deal with bit perfect? Does upsampling increase latency that much? If you are just listening and not trying to capture live sound, does it really make a difference?

I'm asking because I cannot find the term "bit perfect" in FallenAngel's "Before asking about digital audio: READ this Digital Audio Primer"

 

[krmathis]

Quote:

Originally Posted by grokit /img/forum/go_quote.gif Not to hijack or be disrespectful, but what's the deal with bit perfect? Does upsampling increase latency that much? If you are just listening and not trying to capture live sound, does it really make a difference? I'm asking because I cannot find the term "bit perfect" in FallenAngel's "Before asking about digital audio: READ this Digital Audio Primer"

Well, this is Head-Fi!

Where the sake of mind is important. Knowing that the audio stream is bit-perfect without loss.

Some are pleased with lossy encoded audio files and probably do not care about bit-perfect playback. While others use lossless encoded files and go for bit-perfect playback to match it.
Each to their own I guess!