Axiom Home Page Forums General Discussion Technical Questions Digital vs. Analog Specs

I could use a little input on the following:

I play the majority of my music from my computer through my receiver (H/K AVR135). I have a Turtle Beach - Santa Cruz sound card which allows me to output in either digital or analog. I wanted to switch over to the digital output of the sound card. I was looking at the Turtle Beach website and it gave the following stats:

Analog (A-A):
-------------
Frequency Response: 10Hz - 120kHz (-3dB)
SNR: 96dB FS A-weighted.
THD+N: (-3dB): < -91dB FS (0.0027%).
Crosstalk: -105dB @ 100Hz

Digital Playback (D-A):
-----------------------
Frequency Response: (-3dB) @ Fs = 48kHz: 10Hz to 20kHz
SNR: 90dB FS A-weighted
THD+N: (-3dB FS): < -87dB FS (0.004%).

Digital Recording (A-D):
------------------------
Frequency Response: (-3dB) @ Fs = 48kHz: 10Hz to 20kHz
SNR: 93dB FS A-weighted
THD+N: (-3dB FS): < -84dB FS (0.005%).
Digital Output: 48kHz PCM audio or Dolby Digital(R) AC-3 for external decoder.
----------------------------------------------------------------------------
For those who wonder:
The SNR on the Analog is 96 dB whereas the SNR on the Digital is 90 dB & 93 dB. The A-A number is always higher (and basically meaningless since it measures only the analog loopback i.e. line in to speakers).

Do the above numbers indicate that the quality is better through the analog output? I just don't get it.

Any ideas?

Jordan

Looks like the board has a different look. Guess I'll have to get used to it.

Yes, to answer your question, it does appear that the analog specs are better. If our analysis is correct, it could be that the analog to digital convertor on your PC board is limiting the digital range.

Just a thought.

I don't know why I find this so confusing.

What would be considered analog in my computer? All the music played back was either ripped from a CD or downloaded - both of which are digital right?

Cheers,

jordan

The other thing that I don't get is - wouldn't the computer just be sending the digital stream to my receiver where the decoding would be done? Wouldn't this be the best way to do it?

Also, my card only outputs 48 Khz. Would there be a noticable difference with one that outputs 96 Khz?

Cheers,

jordan

It all depends on the circuitry of your computer and sound card. I don't think that there is one standard answer as to how the signals are processed and transmitted.

Also from Jordan "I currently have a Turtle Beach Santa Cruz soundcard and am using the SPDIF output to my receiver for all playback. I thought as well that if I used the digital output that it would A) just pass the digital stream to my receiver for decoding, and B) give me the best sound quality.

The reason I wrote is that I read some specs that had me confused.

Second question, what about using a soundcard that uses a 96 hrz stream over a 48? Think it would be worthwhile?

I'm currently storing music as AAC (Apple iTunes) at 256kbs. Should I be thinking about storing my music some other way? "

Hi Jordan,
1) My understanding that by using the SPDIF port the sound card is just acting as an interface, eg you set your software to use SPDIF and a digital copy of the bitstream from the source is sent to the port and all the processing is done by the AV AMP (hence the D-A specs of your AMP matter not your card).

2) I don't really know the answer with what sampling freq to use, but your AMP would need to support it.

3) AAC@256kbs is not bad and perfectly suited to portable music players etc, but you will notice a difference with a good set up (I certainly did as soon as my Axioms arrived and I had to re-rip the lot in a Lossless format). I've decided to keep everything on the HTPC in a lossless format (for me WMA Lossless made sence but there are others), and then covert them on the fly to MP3 when syncing the portable player.

Thanks
Nathan

Sorry for the quick reply above, but I had to take the son to Rugby (FYI they won in the dying seconds).

1) CD Audio (according to the Sony/Philips Red Book) encodes Audio as PCM 2-Channel, 16 Bit, with 44.1kHz Sampling Rate.
2) S/PDIF (Sony / Philips Digital IF) not surprisingly was designed to support transmitting the above spec (can also support DAT which uses 48kHz sampling + some SPDIF implemenations can also support up to 24 Bit).
3)There are many lossless formats/CODECS that aim to allow you to save the PCM stream in a file format that allows the orginal PCM stream to be faithfully recreated such as (from wikipedia):
* Apple Lossless - ALAC (Apple Lossless Audio Codec)
* Direct Stream Transfer - DST
* Free Lossless Audio Codec - FLAC
* Meridian Lossless Packing - MLP
* Monkey's Audio - Monkey's Audio APE
* RealPlayer - RealAudio Lossless
* Shorten - SHN
* TTA - True Audio Lossless
* WavPack - WavPack lossless
* WMA Lossless - Windows Media Lossless

So for quality playback you are best playing either the original CD or a Lossless version of it, and trasmitting the 2Chn, 16Bit, 44.1mHz PCM stream over S/PDIF to your AV AMP for decoding (presuming it has better DACs - which I am sure it would!).

Likewise, I also use the S/PDIF connection to transmit the Digital Audio Steam from DVD's rips etc to my AV AMP (for decoding) as noted by wikipedia: "Another common use for the S/PDIF interface is to carry compressed digital audio as defined by the standard IEC 61937. This mode is used to connect the output of a DVD player to a home theater receiver that supports Dolby Digital or DTS surround sound."


Thanks
Nathan

Also a bit more info I found on sampling rates (also wikipedia):

In digital audio, common sampling rates are:
8,000 Hz - telephone, adequate for human speech
11,025 Hz
22,050 Hz - radio
32,000 Hz - miniDV digital video camcorder
44,100 Hz - audio CD, also most commonly used with MPEG-1 audio (VCD, SVCD, MP3)
47,250 Hz - world's first commercial PCM sound recorder by Nippon Columbia (Denon)
48,000 Hz - digital sound used for digital TV, DVD, DAT, films and professional audio
50,000 Hz - first commercial digital audio recorders from the late 70's from 3M and Soundstream
50,400 Hz - sampling rate used by the Mitsubishi X-80 digital audio recorder
96,000 or 192,400 Hz - DVD-Audio, some LPCM DVD tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD-DVD (High-Definition DVD) audio tracks
2.8224 MHz - SACD, 1-bit sigma-delta modulation process known as Direct Stream Digital, co-developed by Sony and Philips

The S/PDIF specs says allows it to support different sampling rates but I have no idea if each implementation can support all of them.

While we're on the topic of preferred MP3 bitrates and whatnot, I might as well throw my own question in here...

What is everyone's opinion on variable bitrate MP3 encoding. Most of my music is encoded with LAME at "alt-preset-standard" (now "V 2"). Recently, I've started encoding at "alt-preset-extreme" (now "V 0"). According to Hydrogenaudio, "alt-standard" and better should be near-lossless quality.

Is this all a bunch of hooey, and I'll be wanting to re-encode everything as lossless when I get my Axioms? Anyone have a clue which frequencies you end up losing when you encode MP3s at these bitrates?

Just a matter of where you draw the line... for most computer/micro system/car speakers, 128kbps should be fine... 192/256 is a pretty basic setting for mix of compression and quality... and most encoders top out around 320. Personally I only use MP3s in the car and I encode at 256, more than good enough when coupled with road noise and the basic bad acoustics of an automobile.

Bren R.

Many thanks for your replies Jmone.

I think that I will stick with the sound card that I have then for now. I think that it is doing everything that I need it to for the moment.

I will also play around with some lossless formats to see what the difference is like when the M60's arrive. Only 5 more weeks.

J