ASIO Recording 24 bit (48 or 96kHz) A2 Platinum

I have read at kX Project that the Audigy 2 drivers from July 2004 enables 24 bit/96 kHz ASIO recording with the A2 ZS series. Can I get the same with the A2 Platinum eX? So far I have been using kX drivers with good results, but only with 6 bit and 48 kHz.
Thanks for someone to answer!
Sture in Sweden

"The SB Audigy ASIO 24/96 is only available with the Audigy 2 Platinum EX which uses ASIO 2.0 driver and is fixed on 24-bit/96khz. When using Cubasis you should always have one of these drivers selected and not the Multimedia or the directX drivers. You can adjust the buffer latencies from the ASIO control panel in the Audio System setup from the default 50ms up to 500ms and down to 2ms."
taken from knowledge base
search "ASIO recording" pick the top article

Similar Messages

Audigy 2 ZS 24/96 ASIO Recording

INTRODUCTION
Is Audigy 2 ZS capable of 24/96 asio recording? Yes, it is. You will have analog mix (line in) and vst soft synths available for recording, listening and export as audio mixdown. The sound is crystal clear and you'll have a fine 24 bits and 96 khz song. If you want to have real professional recording you should have a soundcard with professional DACs and so you should consider buying an EMU card.
HOW TO DO IT? I'll use Cubase Sx 1.0.5 as reference. The names may differ but the main process is similar for newer versions of Cubase.
Open Cubase SX and under Devices - Devices Setup - VST Multitrack choose SB Audigy 2 ZS 24/96 as your Asio Driver. Untick "Release ASIO Driver in Background" and "Direct Monitoring", select a useful latency in Control Panel (less than 10 ms is good), hit Ok, Apply and Ok again.
Open a new project (empty) and under Devices - VST Inputs activate only Analog Mix (remap if asked). You're almost ready to rock.
AUDIO
Add an audio track (mono or stereo) and select your input (analog mix).
Now, check your Creative Mixer inputs: you must have Line In unmuted (and at 58% for best sound quality) and you must select Analog Mix at 50% (0 db) in your Rec slider. Check your Audio track in Cubase if you have monitoring signal. You can activate the track monitoring button for listening to any Vst Effects that you want to apply in your track (like reverb, delay...). Arm your track for recording and hit the record button (or * tab) and you'll have your sound recorded (guitar, voice...).
VST SYNTHS
Add a midi track, select your VST instrument (under Devices) and, in your midi track, choose I/O Midi SB Audigy 2 ZS as your Midi In and your VST Instrument as your Midi Out. Don't forget to enable Midi Thru Active under File - Preferences to have sound in your midi track.
After recording your song you can export your audio mixdown and you're done. Export it as Stereo Interleaved and include Automation and Effects.
SOUNDFONTS
You'll not be able to record them to audio, or listening to them when you export your audio mixdown.
VERIFYING RATES
Open your Creative Media Player and open any audio file you've record in Cubase (you''l have it in your Cubase project folder) and you'll see that it's a native 24/96 audio file.
Hope this will help everyone who wants to make good homerecording, without having to spend a lot of money in a professional board.
Message Edited by gemada on 11-04-2005 10:26 PM

Darkk_Nights wrote:
Thanks Jutapa!
I thought the file format were automatically set to 24-bit when the ' SB Audigy 2 ZS 24/96' ASIO driver is choosen. How can I set it to 24-bit?
And another question: why do you said that none of the Audigy series cards are good with 16-bit/44.1 kHz quality? In fact, what would be the difference between a recording made with an Audigy and with some others professional audio cards?
Again, thanks a lot!!!
Cubase SX --> Recording bit depth and SR can be set on Project settings.
1st, Creative Soundblaster cards are not pro cards but just consumer cards, intented for multimedia and gaming use. Creative pro cards comes from EMU brand. This information comes from Creative web pages.
Read this Creative Audigy2 ZS Sound Card Review : http://www.digit-life.com/articles2/creative-audigy2-zs/index.html
Then read couple of US$150 more Pro card reviews
- http://www.digit-life.com/articles2/esi-julia/index.html
- http://www.digit-life.com/articles2/audiotrakmaya44mkii/index.html
Then read couple Pro audio interface reviews
- http://www.digit-life.com/articles2/lynxtwo/index.html
- http://www.digit-life.com/articles2/multimedia/esi-maxioxd.html
Do you now see some of the differences?
jutapa

I recorded 24 bit audio. need to change settings to 24 bitrate

I recorded 24 bit and it sounds terrible. I'm assuming I need to change the bitrate to 24 in FCP. How do I do that. Footage is from
Red One Camera but I'm not importing footage yet. I'm just checking the sound files and they sound terrible.

There is a lot of noise. I did another test and changed it back to 16 bit and it was fine. I played it in Quick time and it did the same thing. I was told I need to change my settings in fcp to 24

Audigy 2 zs MIC ASIO recording problem?

I'm using Sonar 4 to simply record some voices through a micphone. I set the the "Creative ASIO stereo mic in L" as my input and i used a software "compressor" and "reverb" as my pre effects. It's working very well when i start monitioring. So that i can heard wet sound and record dry sound at the same time. But the problem occurs when i click the red "ARM" button and ready to record, i found that the input level are getting much greater and overloaded. I tried a lot but it seemed as if there is no way to adjust the input level of micphone under ASIO recording mode. The pre compressor seemed useless. Is there a way to record a limited or compressed micphone input? Now i have to record in another track and set the input as "Creative ASIO stereo mix in L". But in this way the background music has been recorded aslo. I'm getting mad. I don't know do i need a hardware amp or compressor for micphone.

make sure your have set up microphone in sound option and not "What you hear"Double click the Creative sound menu buttom right, and make sure its microphone thats in the far right?Put the volume to the max to might be that also, even try to use the 20db boost

X-Fi - Selecting bit depth when ASIO recordi

When using the X-Fi with the Creative ASIO driver (in 3rd party applications), how do you select the bitdepth 6 or 24 when recording? The ASIO driver only gives access to the soundcard latency. I am missing something obvious?
digifish
Message Edited by digifish on 0-20-2008 06:54 PM

I can answer this for you.... I'm afraid it wont work from what I've read on other forums in Vista.
First of all you need to leave the creative software alone as that never works, then you do it by right clicking on the speaker icon etc to do what I am doing [url="http://www.youtube.com/watch?v=e5qa-HODWbc">here[/url].
I don't think you will see What U hear in there tho.
Have a read of this massi've thread [url="http://forum.thinkpads.com/viewtopic.php?t=52527&postdays=0&postorder=asc&sta rt=20">here[/url]?post by ]usenet7.
Re: Creative X-Fi "stereo line-in"
Don't bother getting the card if you have Vista. There is no stereo mix or other like capabilities.
Ive spoken to that guy, and I can tell you what he says is correct.
Now the really bad news.. I don't know of ANY Vista laptops that can do what I have in that video either using their onboard sound or any add on card you can buy.

The people at places like singsnap are tearing their hair out over this.

X-Fi - Selecting bit depth when ASIO recording

When using the X-Fi with the Creative ASIO driver (in 3rd party applications), how do you select the bitdepth 16 or 24 when recording? The ASIO driver only gives access to the soundcard latency. I am missing something obvious?
digifish

Unfortunatly thats the way it is with soundblaster cards and vista drivers at the moment. There are a few threads open with people that have been complaining about this for months.Your 3 choices are; run windows xp, deal with the lack of driver features at the moment, or get a different brand of soundcard.

Audigy 4, difficulty with 24/96 ASIO recording

I just purchased an Audigy 4 (SB0610, not Pro) with the intention of using it fir 24-bit recording with n-Track Studio (24-bit). When I'm in n-Track and select the Audigy 4 24/96 ASIO [D000] drivers, I can only get input from the left side of the line-in. Furthermore, when I make certain changes to audio settings in n-Track, I will lose even that input until I use the surround mixer to disable / re-enable the line input.
Is this *supposed* to work? Or am I barking up an incorrect tree?
The WDM driver does work and give me both left and right channels, but I'd like to use the ASIO driver if possible to reduce latency.
So far, I've tried changing my recording source in the mixer. I've tried disabling all loaded programs and unnecessary system services (via msconfig) and I use the auto update feature to update the Creative drivers. The update process itself was a little wierd because it didn't install the Audigy 4 drivers (only the audio console app) until I dug in to the installed programs and ran the driver installation manually.
It seems that if I change my speaker configuration to anything less than 5.1 (2.0, 2.1 etc), then I lose even the left channel. 6.1 and 7.1 give me the same results as 5.1, only left channel and no right.
I've also disabled non-essential system hardware to free up IRQs. I've even replaced the motherboard with an identical model (I found I had bad caps.)
Anyway, I can't seem to get the 24/96 ASIO under control. I also seems unstable as it will trap n-Track from time to time.
The mobo is EpoX 8KHA+, AMD Athlon 1900+, Windows XP SP 2.
Thank you greatly for any help!!!!!!!!

Thank you very much for the reply. The relevent parts seem to be:
(Note that in 24 bit 96KHz mode the inputs are limited to the analog input pairs.)
I'm in 24/96 mode because I picked the 24/96 ASIO driver for both recording and playback in n-Track Audio Devices, correct?
After selecting that driver, I tried to select a sample rate of 96000 in n-Track, but no radio button was selectable. I assume that's because the driver is determining the sample rate, correct?
• ASIO inputs 1 and 2 come from the Sound Blaster Audigy 2 eX Drive Line In 2 stereo input;
!! My NOTE: AnalogMix(Line-In/Aux-In/analog CDAudio/PC (W-U-H)/TAD/...)
In the n-Track settings, I only have one selectable input "SB Audigy 4 ASIO [D000]." Is it safe to assume that this represents the "AnalogMix" of all those physical inputs? If so, them I'm doing the right thing and I still don't know what's wrong. (BTW, what's W-U-H?)
This also may be relevent:
• ASIO inputs 15 and 16 come from the Sound Blaster Audigy 2 eX Drive Line In 1/Mic In stereo input;
• ASIO inputs 17 and 18 come from the Sound Blaster Audigy 2 eX Drive Line In 3 stereo input;
Are these the "analog input pairs" you referred to above? If so, are "Line in 1" and "Line in 3" the left and right channels on the blue connector on the back of my card? It seems kinda hokey to split a stereo signal across two ASIO channels, one of which is not selectable.
I'm not getting it yet. A block diagram would be very handy. Please help me understand.
Thanks again.

ICD-p320 Recorder 64 bit

Long time ago I purchased a digital voice recorder Sony ICD-P320. Barely used it, but trusting in SONY I decided to keep it instead of returning it to the store.
Years go by and I need it now. I kept the installation CD and everything else just to find out that I can not install it in my Windows Vista Home Premium computer (64 bit) because SONY does NOT support the 64 bit driver, while it does support the 32 bit. I've spend >2 hours browsing the internet and trying different tricks (Windows compatibility, etc) to no avail.
Has somebody developed the driver for the 64 bit Windows Vista?
Why do I have to lose my money and a piece of HARDware just because of lack of SOFTware?
Thanks

Probably let me just add one more comment since I think it's very unlikely that somebody developed the driver for the Sony recorder that I happen to have (which is a good recorder, has just been made obselete due to software incompatibility).
For my comment to make any sense to you, please watch the following short and fun documentary called The Story of Stuff (http://www.storyofstuff.com/sos.php). Even if you just watch the documentary without reading my comment, that will be enough.
As it turns out, most of the industry and as a prime example, the electronics and software industry, has been relentlessly plunged themselves and consumers along into a new era called CRAPOCRACY or the consumption of CRAP. This is characterized for developing new and newer pieces of electronics that although some of them do represent a significant improvement (digital vs analogic recorders), are made deliberately obsolete by software incompatibilities and the protection of the so called Property Rights. The hardware could be intact and be very useful indeed but because of protection of 'intellectual property', companies render a product that is virtually crap. Companies do make money this way by halting the competition and stimulating faithful consumers into purchasing newer products to make electronic pieces compatible again. Overall, they are winning. But at a very high price. At the expense of overconsuming the natural resources while keeping everybody stuck with useless stuff.
Take a look at the Carbon Dioxide concentration in the atmosphere surpassing the 350 ppm considered the highest safe margin by the Kyoto protocol. We're at 385 ppm right now!!. We're collapsing!!. And all we hear from politicians is that we need to stimulate the economy!!.
But let me remind you that even during the last day, Sony will publish a list of products it does no longer support. In short, we live in a society deemed to profound ignorance, in a world full of useless crap and by that time we rock bottom, we'll be still be lacking the technology to conquer new worlds (outside planet crappy Earth), especially since the Universe is expanding and becoming unreachable. We must thank Sony and others who have made a commitment to remain stuck in developing more of same newer sh1t that is supposed to replace older, discutinued and incompatible sh1t. Another world is possible, that's what I believe.

Capturing 4 channel recording (12 bit setero 1,2 mix) into FCP4

Having recorded from the on board mic and radio mic as a 12 bit 4 channel mix on my Canon XL1s, I cannot capture the audio mix. FCP only captures the stereo 1 (on board mic) and not the mix. It recorded well on the camera and I heard the mix on the review of the tape, but I'm struggling to get the mix on FCP (i.e. all four channels).
I've invested £500 in radio mics and the adapter which all work well. Come on you FCP buffs! Other forums have yet to produce the goods. Many thanks in anticipation.
Mark
G5 Mac OS X (10.3)

I don't know the audio setup of your camera, but if you take it as a straight audio term, a 12 bit 4 channel mix means you recorded one 12 bit audio track that consists of a mix of the 4 inputs on the camera.
I would hope that's not what it really means.
That being said, you can't capture all 4 at once. You have to capture in pairs. Capture your video with channels 1 & 2, then create a new clip with the same in & out points, make it audio only, and change the audio settings in FCP (and maybe your camera, too) to 3 & 4 and capture that clip.
Now put them on the timeline together and link them using Modify>Link.

Recording extra bits into MIDI regions DRIVES ME MAD - please help!

I hate Logic sometimes. (Logic Pro 8)
I record a simple part (region) using an AU plugin - a basic beat let's say. I play it in with a keyboard.
It's a 2 bar region and I then want to add a cymbal into the region while in 'piano roll' view (so I can see where to add it) - so I hit record and 1) my view of what's already been recorded goes blank and 2) Logic creates a NEW region!
Is there not a way that when in Piano Roll view - if you want to record extra MIDI stuff into a region - you can start recording and can still see what you already recorded in that region, and for your new recording to AUTOMATICALLY merge in with the existing region?
This drives me crazy and i've never figured it out - and it really stands in the way of me actuallly getting anywhere with Logic. I just want a region to stay as ONE region which I can keep adding bits to without creating loads of regions and I'd really like to be able to SEE what's already there as I add bits!
Can anyone help pls?

Florence wrote:
thanks a lot guys - BUT is there also a way for me to see the MIDI notes i record AS I actually play them in? (rather than when the looped area cycles round again) This happens in Reason and most other music apps - just annoys me it doesn't in Logic!
Don't know why that doesn't happen, I imagine it's the way Logic holds incoming MIDI data in a buffer instead of committing to a note value, but I don't know. Just tried the same thing in OEM version of CubaseLE4 and played notes became visible in real time.
pancenter-

Noisy sound data when recording 16 bit. Anyone experience this?

I'm working on a program that displays a spectrogram and it works just fine with 8-bit audio, but not so well with 16-bit. In the 16-bit data there's an incredible amount of noise that isn't present in the 8-bit data. Oddly, the noise is present across the entire range of frequencies I'm reading. If it makes a difference, I've only tested this on Vista. Here is an abbreviated code snippet:
ArrayList<double[]> wave = new ArrayList<double[]>();
int sampSize = audioFormat.getSampleSizeInBits();
numBytesRead = targetDataLine.read(data, 0, data.length);
wave.add(new double[dataSize]);
if (sampSize == 16) {
 for (int i = 0; i < numBytesRead / 2; i++) {
 wave.get(wave.size()-1) = (double) ((data[2*i+1] << 8) | data[2*i]);
} else if (sampSize == 8) {
for (int i = 0; i < numBytesRead; i++) {
wave.get(wave.size()-1)[i] = (double) data[i];
//perform fft on wave data

Alright. I've created an "sscce" as suggested, and managed to keep it just under 20KB. It's pretty self explanatory but I will describe it briefly:
The program graphs the waveform of data read from a microphone. Two graphs are displayed. The top graph displays the data read in 8-bit form, and the bottom graph reads the data in 16-bit form. I wasn't sure if it was possible to read in 8-bit data and 16-bit data from the same source simultaneously, so you have to toggle between them. Clicking in the graph area will pause/unpause that graph, allowing the other graph to run. Pausing both graphs allows for the data to be compared side-by-side.
This image illustrates my problem: http://yfrog.com/3d8vs16graphj. The data was recorded during silence, so what you see is purely noise. The 16-bit data clearly has plateaus that aren't present in the 8-bit data. I believe these "plateaus" are the cause of the broad spectrum of noise in my fft results. If anyone is interested, please test my code by copying and pasting directly into a new project:
package sscce16bit;
import java.awt.Color;
import java.awt.Dimension;
import java.awt.Graphics;
import java.awt.GridBagConstraints;
import java.awt.GridBagLayout;
import java.awt.Image;
import java.awt.Insets;
import java.awt.Point;
import java.awt.Rectangle;
import java.awt.Toolkit;
import java.awt.event.MouseEvent;
import java.awt.event.MouseListener;
import java.awt.event.WindowAdapter;
import java.awt.event.WindowEvent;
import java.awt.geom.Rectangle2D;
import java.util.ArrayList;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.TargetDataLine;
import javax.swing.BorderFactory;
import javax.swing.JFrame;
import javax.swing.JPanel;
import javax.swing.SwingWorker;
public class sscce extends JPanel {
 static MainThread mainThread;
 static final int gh = 100;
 static final int gw = 300;
 static final int ch = 40;
 static final int labw = 60;
 static final int cw = gw;
 static final Dimension graphSize = new Dimension(gw, gh);
 static final Dimension labelPSize = new Dimension(labw, gh);
 static final Dimension controlsSize = new Dimension(cw, ch);
 static final Dimension panelSize = new Dimension(gw + labelPSize.width + 20,
 2 * (labelPSize.height + ch));
 static GraphPanel graph8;
 static GraphPanel graph16;
 static ControlsPanel cPan;
 int procCounter = 0;
 int curArrayPos = 0;
 TargetDataLine targetDataLine8;
 TargetDataLine targetDataLine16;
 //double maxAmpShown = -1.0d;
 //double MAX_AMP = 0.0d;
 ArrayList<double[]> wave = new ArrayList<double[]>();
 boolean pauseProc = false;
 //graph display controls
 int tickMarkLen = 8;
 int numOfLabels = 5;
 //audio format
 AudioFormat audioFormat8;
 AudioFormat audioFormat16;
 int dataSize = 1024; //buffer size = 2^10 (line buffer size is 4000 typically);
 final int dataSampleSize = dataSize;
 final float sampleRate = 8000.0F; //8000,11025,16000,22050,44100
 final int sampleSizeInBits8 = 8; //8 or 16
 final int sampleSizeInBits16 = 16;
 final int numofchannels = 1; //1 or 2
 final boolean signed = true;
 final boolean bigEndian = false;
 public sscce () { //constructor
 audioFormat8 = new AudioFormat(sampleRate,
 sampleSizeInBits8, numofchannels, signed, bigEndian);
 audioFormat16 = new AudioFormat(sampleRate,
 sampleSizeInBits16, numofchannels, signed, bigEndian);
 this.setPreferredSize(panelSize);
 this.setMinimumSize(panelSize);
 this.setMaximumSize(panelSize);
 this.setBorder(BorderFactory.createLineBorder(Color.black));
 this.setLayout(new GridBagLayout());
 GridBagConstraints gbc = new GridBagConstraints();
 gbc.insets = new Insets(0, 0, 0, 0);
 gbc.anchor = GridBagConstraints.CENTER;
 graph8 = new GraphPanel(sampleSizeInBits8,
 audioFormat8, targetDataLine8, true);
 gbc.gridx = 0;
 gbc.gridy = 0;
 gbc.gridwidth = 1;
 add(graph8.labP, gbc);
 gbc.gridx = 1;
 gbc.gridy = 0;
 gbc.gridwidth = 1;
 add(graph8, gbc);
 cPan = new ControlsPanel();
 gbc.gridx = 1;
 gbc.gridy = 1;
 gbc.gridwidth = 1;
 add(cPan, gbc);
 graph16 = new GraphPanel(sampleSizeInBits16,
 audioFormat16, targetDataLine16, false);
 gbc.gridx = 0;
 gbc.gridy = 2;
 gbc.gridwidth = 1;
 add(graph16.labP, gbc);
 gbc.gridx = 1;
 gbc.gridy = 2;
 gbc.gridwidth = 1;
 add(graph16, gbc);
 mainThread = new MainThread(this);
 mainThread.execute();
 } //end constructor
 class ControlsPanel extends JPanel {
 ControlsPanel() {
 this.setPreferredSize(controlsSize);
 this.setMinimumSize(controlsSize);
 this.setMaximumSize(controlsSize);
 @Override
 public void paint(Graphics g) {
 String pause8, pause16;
 String zoom8, zoom16;
 Rectangle2D textR[] = new Rectangle2D[6];
 if (graph8.pauseGraph) {
 pause8 = "paused";
 } else {
 pause8 = "running";
 if (graph16.pauseGraph) {
 pause16 = " paused,";
 } else {
 pause16 = " running,";
 zoom8 = " " + Double.toString(graph8.MAX_AMP / graph8.maxAmpShown) +
 "x zoom";
 zoom16 = " " + Double.toString(graph16.MAX_AMP / graph16.maxAmpShown) +
 "x zoom";
 String text[] = new String[] { "8bit: "+ pause8 + zoom8,
 "16bit: " + pause16 + zoom16 };
 for (int i = 0; i < text.length; i++) {
 textR[i] = g.getFontMetrics().getStringBounds(text[0], g);
 textR[0].setRect(0, textR[0].getHeight(),
 textR[0].getWidth(), textR[0].getHeight());
 textR[1].setRect(cw / 2.0d, textR[1].getHeight(),
 textR[1].getWidth(), textR[1].getHeight());
 for (int i = 0; i < text.length; i++) {
 g.drawString(text, (int) textR[i].getX(), (int) textR[i].getY());
class GraphPanel extends JPanel implements MouseListener{
double maxAmpShown = -1.0d;
double MAX_AMP = 0.0d;
boolean pauseGraph = false;
AudioFormat audioFormat;
CaptureThread capThread;
TargetDataLine targetDataLine;
LabelsPanel labP;
ArrayList<Integer> labels = null;
boolean setLabels = true;
boolean stopped = true;
Image bImage;
Graphics bg;

Novice requires bit of help with K8N Platinum

Hello, I have just upgraded my PC with this motherboard and processor and I require a little assistance,
first of all how do I tell if the memory is running in dual channel mode? I can see the full 1GB but am not sure if I'm getting full potential out of it.
next I'm using a Samsung SATA II drive but when I do a speed test in the device manager in control panel I get a resulting bust speed of about 133 out of 300 max, is this normal or is the drive running in SATA I mode?
Many thanks

Quote from: Lakes_Puma on 15-May-06, 16:57:16
Hello, I have just upgraded my PC with this motherboard and processor and I require a little assistance,
first of all how do I tell if the memory is running in dual channel mode? I can see the full 1GB but am not sure if I'm getting full potential out of it.
if you have the memory set in the first two slots (green and purple) it should run dual-channel. you can check with CPU-Z or when the system boot, check if the memory is listed as 128bit (dual-channel) or 64bit (single-channel)
Quote from: Lakes_Puma on 15-May-06, 16:57:16
next I'm using a Samsung SATA II drive but when I do a speed test in the device manager in control panel I get a resulting bust speed of about 133 out of 300 max, is this normal or is the drive running in SATA I mode?
Many thanks
the 300 is the maximum theoretical speed of the SATA controler. in reality, you will get about the same performance as a PATA harddrive. all the SATA I and II specs are for marketing purposes only, and there is no harddrive in the world that can get to 300MBps transfer speed at the moment. so the 133 burst speed is very good.

Audigy 2 ZS 24-bit/96kHz Recording support

Hi
According to Creative's website,on Feb 2004,
only SBAudigy2Plat.Pro had --> ASIO 2.0 Low Latency Multi-Track Recording Support (24-bit/96kHz)
See http://web.archi've.org/web/2004028023909/http://www.soundblaster.com/products/Audigy2ZS_platinum_pro/compare.asp
But now, after Audigy4's release, the website's comparision chart shows all the members of the Audigy2 family can do 24bit/96kHz ASIO 2.0 Recording.
See:
http://www.soundblaster.com/products/Audigy2ZS_platinum_pro/compare.asp
Please clarify this ?
Were any new driver released by Creative to add support for 'ASIO 2.0 Low Latency Multi-Track Recording(24-bit/96kHz)'
for Audigy 2 ZS, as well as Audigy 2 ZS Platinum ?

-Q,
This has been added to the drivers. Make sure you are running the latest drivers and this should update you to ASIO 2.
Jeremy

Bit-Match Recording

? Hi folks,
I am having a problem with Bit-Matching Recording... I have tried to record from both my mic pre amp that outputs digitally and from another PC both via optical cables. The problem is when I set the output of my preamp or 2nd computer to 96khz the Bit-Range monitor shows the correct sample rate and when I record (from the Console Launcher) in Bit-Match mode the file records at 96khz, however no matter which application I play the file back with the file plays back really slow. Bit Match recordings at 44. and 48khz are fine.
Also does anybody know if the bit match option works with ASIO or other audio apps? When I have Bit Match option enabled all my applications (Vegas, Acid etc) still allow me to record at any bit rate. I know with other interfaces the clock should be locked to the digital input and this cannot be changed in the application. So does the card still perform sample rate converstion to the digital input?
Cheers!?

mmm ok that really has nothing to do with what I am talking about
I am talking about the sample rate of digital audio coming into the Optical input and the realtion between Bit-Match (which is supposed to set the card to record to the same sample rate as the audio coming into the optical input) and the Sample Rate Conversion on the X-Fi chip. There is something wrong with the card recording a sample rate of 96khz from a 96khz source. Also other applications allow recording from the optical in at any sample rate even though Bi - Match recording is enabled, and I don't think it should be.
Cheers

Audigy Platinum only records at 48k 16 bit in Sound For

Can someone please tell me what i'm doing that's so stupid? I'm trying to use Sound Forge to record my DJ sets through my Audigy Platinum, but for some reason Sound Forge will only let me record at 48k 6 bit sample rate, and the monitoring is almost impossible - most of the time I get terrible feedback destroying my ears! If I then save something I've recorded as a 44.k wav file or a mp3 file, the sound blaster wont play it at all.
I've moved the Audigy from an older PC that died on me, and I didn't have the original disc (from 4 years ago) so I downloaded the drivers from the web, but it doesn't seem to be working too well.
Any suggestions on what I might be doing wrong?
Cheers.
Simon

szimbler wrote:
Ok, I have some more info in this. I can get Soundforge to work when it uses the Windows Sound Mapper, but not the Creative Asio driver. DOes this help? Does it matter which one Sound Forge uses?
Please help!
Your card supports ASIO only @ 6-bit/48 kHz (hard locked).
Are there other driver modes listed (MME Audigy, DS Audigy, DirectX)?
If 6/48 with ASIO is not enough, then use DS Audigy if possible.
jutapa

ASIO Recording 24 bit (48 or 96kHz) A2 Platinum

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