[Rizlaw]

Question:
 
Assuming a computer (Windows, Mac or Linux) properly configured for bit-perfect digital audio output with superior grade parts and power supply, are there any (negative or positive) quantitative differences which would ultimately affect the final analog sound signal heard, between the following two digital signal paths:
 
      (a) ALC889 coax s/pdif digital out signal from a motherboard sound chip to an external DAC (i.e., W4S DAC2) > AMP > headphone/speakers;
 
      (b) coax s/pdif digital output from a PCIe Asus Xonar STX card to an external DAC (i.e., W4S DAC2) > AMP > headphone/speakers.
 
I ask this question, because, although there may (or may not) be quantitative differences, in my own experience, I hear no negative qualitative differences either through headphones or speakers using FLAC 16/44.1, 24/96 and 24/192 source material. This suggests to me that if one is simply outputting an unaltered digital signal to an external DAC > AMP > headphone/speakers, there is really no need for a high quality plug-in sound card as the real audiophile work is being performed by the external DAC and AMP.
 
 
 

 

[vitaminc]

Interested to know as well!

 

[MaciekN]

From what I've read in ALC889's datasheet it can not output bit perfect signal at 44.1kHz sampling rate.

 

[trog]

Hmm correct me if i am wrong but as far as digital out is concerned onboard mobo sound or discrete soundcard matters not right? The same magic occurs with the external DAC regardless me thinks hehe

 

[MaciekN]

I'm using mobo soundcard ALC883 to output digital signal to an external DAC, and there was a definitive improvement when I borrowed EMU-0404 and used it instead. Quality of soundcard even with digi out matters but it boils down to how well given DAC can clean up trash coming from PC. So you won't probably know until you test it

 

[Rizlaw]

 
MaciekN,
 
My personal experience says you are incorrect on this point (ALC889 can't output bit perfect signal @ 44.1kl). The Realtek PDF I downloaded from Realtek on the ALC889 is 78 pages long, and while I'm no EE, page 2 of the Realtek PDF states:
 
Quote:

ALC889
(PN: ALC889-GR, ALC889DD-GR)
7.1+2 CHANNEL HD AUDIO CODEC WITH
CONTENT PROTECTION
DATASHEET
Rev. 1.0
08 July 2008
Track ID: JATR-1076-21

 
2.1. Hardware Features
High performance DACs with 108dB signal-to-noise ratio(A-weighting)
High performance ADCs with 104dB signal-to-noise ration (A-weighting).
Meets Microsoft WLP3.10 and future WLP audio requirements
Ten DAC channels support 16/20/24-bit PCM format for 7.1 sound playback, plus 2 channels of
concurrent independent stereo sound output (multiple streaming) through the front panel output
Three stereo ADCs support 16/20/24-bit PCM format, multiple stereo recording
All DACs supports 44.1k/48k/88.2k/96k/176.4k/192kHz sample rate
All ADCs supports 44.1k/48k/88.2k/96k/176.4k/192kHz sample rate
Primary 16/20/24-bit SPDIF-OUT supports 32k/44.1k/48k/88.2k/96k/192kHz sample rate
Secondary 16/20/24-bit SPDIF-OUT supports 32k/44.1k/48k/88.2k/96k/192kHz sample rate
16/20/24-bit SPDIF-IN supports 32k/44.1k/48k/96k/192kHz sample rate
All analog jacks (port-A to port-G) are stereo input and output re-tasking
Port-A/B/C/D/E/F built in headphone amplifiers
Port-B/C/E/F with software selectable boost gain (+10/+20/+30dB) for analog microphone input
High-quality analog differential CD input
Supports external PCBEEP input and built-in digital BEEP generator
Software selectable 2.5V/3.2V/4.0V VREFOUT
Up to four channels of microphone array input are supported for AEC/BF applications
Two jack detection pins each designed to detect up to 4 jacks plugging
Supports analog GPIO2 to be jack detection for CD input which is used as 9th analog port
Supports legacy analog mixer architecture
Up to 3 GPIOs (General Purpose Input and Output) for customized applications. GPIO0 and GPIO1
share pin with DMIC-CLK and DMIC-DATA.
Supports mono and stereo digital microphone interface (pins shared with GPIO0 and GPIO1)
Supports anti-pop mode when analog power AVDD is on and digital power is off.
Content Protection for Full Rate loss-less DVD Audio, Blue-Ray DVD and HD-DVD audio content
playback (with selected versions of WinDVD/PowerDVD)
Hardware Zero-Detect output volume control
1dB per step output volume and input volume control
Supports 3.3V digital core power, 1.5V or 3.3V digital I/O power for HD Audio link, and 5.0V
analog power
48-pin LQFP ‘Green’ package
 

 
Perhaps you can state what (and where) you are reading in the "datasheet" that supports your claim? Perhaps my W4S DAC-2 display is wrong when it reads that it's locked on a 44.1k 16 bit signal?
 
Quote:

From what I've read in ALC889's datasheet it can not output bit perfect signal at 44.1kHz sampling rate.

 
 

 

[MaciekN]

Hmm, thats interesting.
 
From page 18 of the pdf datasheet version 1.0:
"24MHz bitclock sourced from HDA controller and connecting to all codecs
48kHz signal used to synchronize input and output streams. It is sourced from HDA controller and is connecting to all codecs."
 
It may support 44.1 at it's input, because how I understand it when the data is recieved it gets upsampled to match the clock.
 
Now, if it's using only 24MHz clock then:
24000000(clock rate)/44100(signal rate) = 544,21768707482993197278911564626. It may be hard to synchronise.
same for upsampled to 48kHz
24000000/48000=500 Looks like there is one signal sample per 500 cycles of clock, easy to synchronise.
 
I am not sure how your DAC can display 44.1, maybe the signal is downsampled somewhere but it looks like ALC889 operates on 48kHz.
 
Do correct me if I'm wrong.

 

[Rizlaw]

As I said, I'm no EE, so I can't comment on what you're saying and whether it's right, wrong or completely irrelevant to the issue of bit perfect output.
 
It was always my understanding, that when the OS and playback software are correctly set for bit perfect digital output, that a computer motherboard DAC (i.e. ALC889) or, for that matter, an internal sound card DAC ( i.e. Xonar STX) simply pass an unaltered digital data stream to an external DAC via S/PDIF or TOSLINK output. Hence, the internal DAC isn't manipulating the data bits in any way, simply passing them, bit perfect, to an external DAC where they are ultimately processed and converted to an analog signal.
 
I do know that both my Linux and Win7 setups are set to correctly pass bit perfect digital output and do.
 
As for my W4S DAC-2, I seriously doubt it's mis-reading the input signal received from the computer's S/PDIF output. It correctly auto switches from 44.1 to 96 to 192 depending on the type of FLAC file being played. It does this whether I use the ALC889 S/PDIF digital out or any of my other sound cards (Card Deluxe 24/96 or modified Asus Xonar STX) S/PDIF outs.
 
 
Quote:

Hmm, thats interesting.
 
From page 18 of the pdf datasheet version 1.0:
"24MHz bitclock sourced from HDA controller and connecting to all codecs
48kHz signal used to synchronize input and output streams. It is sourced from HDA controller and is connecting to all codecs."
 
It may support 44.1 at it's input, because how I understand it when the data is recieved it gets upsampled to match the clock.
 
Now, if it's using only 24MHz clock then:
24000000(clock rate)/44100(signal rate) = 544,21768707482993197278911564626. It may be hard to synchronise.
same for upsampled to 48kHz
24000000/48000=500 Looks like there is one signal sample per 500 cycles of clock, easy to synchronise.
 
I am not sure how your DAC can display 44.1, maybe the signal is downsampled somewhere but it looks like ALC889 operates on 48kHz.
 
Do correct me if I'm wrong.

 
 

 

[MaciekN]

You can not set bit-perfect playback, you can just disable everything that's disturbing it. From the software side you can avoid kmixer in windows or other messy stuff in other OSes but when your soundcard has to output signal in real-time, as is music, then it can not just "send it". It must send samples in correct time intervals, that's why it needs this "clock"; although some DACs buffer that data and clock them on their own which should greatly reduce the importance of soundcard.
 
Maybe I haven't made it clear in my last post, if ALC889 is upsampling stuff to 48kHz then the signal has changed and it breaks the rule of bit-perfectness. This was a common trick, first used by Creative in their first lines of Sound Blasters if I remember well.
 
I too doubt that your DAC is misreading input sample rate but I can not understand how ALC889 could output data at such sample rat. Hopefully someone with more knowledge will comment on that.
 
How do you check the bit-perfectness in your system?

 

[Thunderbolt]

when transfering audio digitally, then it doesnt really matter (as long as your soundcard supports the corresponding sample rates & bit depths)
 
with analogue output, the essence is superior