[ixeo]

Quote:

Originally Posted by Asmo We'll know when someone hears it, technical specs can only give a hint at what is possible.

but if it resamples then its crap right? dont really need to hear that to know it? *assumption*

 

[mikechai]

http://www.soundblaster.com/products/Audigy4pro

 

[Glassman]

Benchmark DAC1, Bel Canto DAC2 and many others do resample, does it hurt someone?? people do resample in software, are they trying to mess up the sound that way?

 

[Glassman]

Audigy4:

Audigy2 ZS:

 

[lan]

The resemblance is uncanny! I wonder if they'll sell the outside box separate for 2 ZS users

 

[Glassman]

you can find even higher resolution photos of the pictures posted.. they evidently did something with the analog stage, it's the only part that looks different on both cards, but the opamps doesn't look like a product of JRC, so no njm2068 grabbed from E-MU fellas

I wonder what are they then.. and another thing to notice, do anyone see two voltage regulators there? I see just one, that would mean those retards in R&D finaly found out +/- 12V opamp's supply is not only cheaper, but a lot better too..

 

[gaboo]

From what I recall of ATX specs, the -12V line is pretty thin (unlike the +12V one), around 1A or so... Maybe that's why soundcards shy away from it.

 

[Glassman]

even less than that, but again, who else besides soundcards use it these days? the specs for PCI bus calls for 500mA on +12V and 100mA on -12V, that's plenty for a few opamps and of course ALL soundcards with opamps on them use +/- 12V..

even older Audigy cards did that, my concern is just with the fact they used +/- 5V regulators from +/- 12V..

 

[Reck45]

What I don't understand about resampling is why are the creative cards so bad for resampling and why is the av710 in high output mode (that resamples) so good?

I'm curious how this is going to stack up against envy chipset based cards...

 

[morsel]

The Benchmark DAC1 upsamples to 192kHz and plays back with 52kHz audio bandwidth. This is good because the waveform is reproduced in the analog domain more accurately at the higher sample rate, then filtered with a greater margin between the sample rate and the filter frequency, which is still well above the limit of hearing of the most golden ears on the planet.

Soundblaster Live and Audigy series resample to 48kHz, which is too close to 44.1kHz and not an even multiple, causing nasty artifacts before the signal ever gets to the D/A converters at the outputs.

Excerpt from the Audigy 4 English link:

Quote:

High Definition Audio Quality for Playback and Recording Playback of 64 audio channels, each at an arbitrary sample rate 24-bit Analog-to-Digital conversion of analog inputs at 96 kHz sample rate 24-bit Digital-to-Analog conversion of digital sources at 96 kHz to analog 7.1 speaker output 24-bit Digital-to-Analog conversion of stereo digital sources at 192 kHz to stereo output 16-bit to 24-bit recording sampling rates: 8, 11.025, 16, 22.05, 24, 32, 44.1, 48 and 96 kHz. Supports Sony/Philips Digital Interface (SPDIF) format of up to 24-bit/96 kHz quality. Selectable sampling rate of 44.1, 48 or 96 kHz Low latency multitrack recording with ASIO 2.0 support at 16-bit, 48 kHz and 24-bit, 96 kHz resolution.

Excerpt from the Audigy 4 Japanese link:

Quote:

Mounting the digital input which does not mind sampling rate conversion With Sound Blaster Audigy 4 Pro, we mount the digital input which does not mind sampling rate conversion, the digital sound recording which is faithful to the original source is possible. The BS/CS tuner, taking in, various digital sources such as CD/DVD player and DAT deck to the computer, you will record & will compile, will try making the original. Through usual sampling rate conversion, the case where you record digitally, the input source be sure to have to adjust the sampling frequency of the software which is recorded,, but as for Sound Blaster Audigy 4 Pro because it corresponds to also the monitoring of sampling frequency of digital input, sampling frequency of the signal which is input from the external equipment is found at the glance.

While this is translated technobabble, there is some room for possible interpretation that the Audigy 4 does not resample. We will find out once the reviewers get ahold of it.

 

[gaboo]

Quote:

Originally Posted by morsel The Benchmark DAC1 upsamples to 192kHz and plays back with 52kHz audio bandwidth.

E-MU 1212m has aproximately the same bandwidth at 192Khz rate. It's been measured to roll off -3dB around 55Khz. It's been confirmed that it uses this fixed anlog filter at all sampling rates.

Quote:

This is good because the waveform is reproduced in the analog domain more accurately at the higher sample rate, then filtered with a greater margin between the sample rate and the filter frequency, which is still well above the limit of hearing of the most golden ears on the planet. Soundblaster Live and Audigy series resample to 48kHz, which is too close to 44.1kHz and not an even multiple, causing nasty artifacts before the signal ever gets to the D/A converters at the outputs.

Unlike the Audigy 2 where host resampling produce audible improvement, resampling in foobar to 192Khz before sending the data to the E-MU 1212m does nothing that my ears can percieve...

Granted, the CS chip that does ASRC in DAC1 has a theoretical time domain bandwidth of 200Ghz, far more than current PCs CPUs can hope to match. But my undestanding is that the main advantage is it's immunity to jitter.

Quote:

While this is translated technobabble, there is some room for possible interpretation that the Audigy 4 does not resample. We will find out once the reviewers get ahold of it.

In view of the recent discovery that E-MU 0404 has a single clock and still measures close to 1212m at 44.1Khz (and some people think it actually sounds better than 1212m), the non-resampling argument isn't very compelling.

 

[Glassman]

Quote:

Originally Posted by morsel The Benchmark DAC1 upsamples to 192kHz and plays back with 52kHz audio bandwidth. This is good because the waveform is reproduced in the analog domain more accurately at the higher sample rate, then filtered with a greater margin between the sample rate and the filter frequency, which is still well above the limit of hearing of the most golden ears on the planet.

that's not a correct statement really, each and every modern DAC has it's own digital interpolation filter at least 8x for 44.1/48, and that's just a filter prior to delta sigma modulator, which runs 128x 44.1/48.. it's clear that resampling prior to DAC does nothing important regarding simplifying analog filter.. however, the fact is that ASRC usually attenuates incomming jitter and it's digital filter is usually better than what's found inside DACs..

the older Benchmark probably use the very same CS parts as the two Audigy cards I've linked images here.. the newer Benchmark use AD1896..

 

[Talonz]

From the translation and the specs, it seems to be saying that the digital -input- does not resample. Nothing is said about the output.