Creative Audigy 2 NX Bit Depth / Sample Rate Prob

This is my first post to this form
Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
Here are my system specs:
HP Pavilion zd 7000
Intel Pentium 4 3.06 GHz
GB Ram
Windows XP Prof. SP 2
Thnx.
-cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

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Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

Audigy 2 NX recording sample rate question

Hi, I have two questions about the sample rate used when recording in the Audigy 2 NX -
When recording an analog signal (Line In), is the actual?sampling rate at which the analog signal is being sampled different from the requested sample rate ? For example, when recording at XX Khz, does the actual sampling of the analog signal really?happen at XX Khz or does it get sampled at a different fixed rate(48khz?) and then gets re-sampled to XX Khz ?
When recording a digital signal (spdif in), if the recording sampling rate specified is the same as the sampling rate of the input signal then does the card record this digital input signal in a bit-perfect fashion ? For example, if the digital input signal is 6bit and 44.Khz, and the recording is also being done at 6bit and 44.Khz then does the card actually record the signal at exactly that rate without modifying the data in any way ? or does it record it at a different rate and then re-sample internally to arri've at the requested recording rate ?
Now I know this card works a little different in terms of re-sampling when the 'Digital Input only' checkbox is checked. So I am looking for the answers to these questions keeping in my mind this option as well.
Thanks!
Message Edited by whatcreative on 02-8-2009 08: PMMessage Edited by whatcreative on 02-8-2009 08: PM

thanks for your replies.
katman,
I checked that thread. Although I wonder if the same thing applies to recording at rates higher than 48khz as well ?as I am more interested in finding out if when recording 96khz does the signal really get sampled at true 96khz or something else ? But, Upsampling a recording (not playback)?to a higher frequency doesn't make sense?because you can't create what you lost by recording at a?lower sampling rate. A test for checking this would be to try and record a signal that has frequency content well above 24khz but less than 48khz. If that content gets captured that means it is indeed recording at 96khz, if it gets lost then its recording at 48khz.
flipflop,
I have opened up the soundcard and messed around with a lot. Here's a list of the chips alongwith their descriptions based on whatever I could find?about them -
Sigmatel STAC9460S - Codec
JRC 4556A - Headphone output opamp
ST MC33078 - Line level output/input dual opamp. (Total 4).
Philips UDA334 - Low Power DAC
Philips ISP58BD - USB device controller
Creative CA086 - DSP
Cirrus Logic EP7309 - High-Performance, Low-Power System-on-Chip Enhanced Digital Audio Interface
ST M29W400BB - 4Mbit Flash Memory
ST LD33 - 3.3v regulator (?)
ST LD8 - .8v regulator (?)
Although the datasheet of the codec suggests that it supports all sampling rates, I'd have to dig deeper to really find out if it is handicapped to only one standard rate for recording - http://www.alsa-project.org/~james/d...tac9460-ds.pdf
Message Edited by whatcreative on 02-2-2009 :35 AM

Determining bit and sample rate of audio file.

Bit of a newbie, and i'm wondering how to determine what the sample rate and bit rate that an audio file has been previously recorded. I'm using Mbox 2 Pro.
Thanks

Open the Audio window and import the file.
cheers
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Creative Audigy and M-Audio Studiophile LX4 Prob

Hi
I build computers but I hang my head in shame at my knowledge of sound cards in my computer I have an Creative Audigy sound card and I have just purchased the M-Audio Studiophile LX4 2. speaker system.
I?m having a problem hooking it up as at the moment the sub is running of the left and right inputs. If I plug the sub input into the digital/audio out like I?m supposed to it doesn?t do any thing. I have CMSS enabled so I know it'ss not that.
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...just use the normal stereo output (green plug) for 2. speakers. Despite the name, there is no separate "sub" input on ANY 2. speakers on the market: they all work with a stereo signal, and the sub channel is separated by a low-pass filter inside the speaker's amp itself.

Bit Depth and Bit Rate

I have a pre recorded mp3 VO. I placed it into a track bed in GB. Clients wants a compressed audio file with bit depth: 16 bit and bitrate: 128kps max, but recommends 96kbps. If I need to adjust the bit depth and bite rate, can I do it in GB? and if so, where? Thanks for any help.

Please be aware that Bit Depth and Bit Rate are two completely different things!
They belong to a group of buzz words that belong to Digital Audio and that is the field we are dealing with when using GarageBand or any other DAW. Some of those terms pop up even in iTunes.
Digital Audio
To better understand what they are and what they mean, here is a little background information.
Whenever dealing with Digital Audio, you have to be aware of two steps, that convert an analog audio signal into a digital audio signal. These magic black boxes are called ADC (Analog Digital Converter) and “on the way back”, DAC (Digital Analog Converter).
Step One: Sampling
The analog audio (in the form of an electric signal like from an electric guitar) is represented by a waveform. The electric signal (voltage) changes up and down in a specific form that represents the “sound” of the audio signal. While the audio signal is “playing”, the converter measure the voltage every now and then. These are like “snapshots” or samples, taken at a specific time. These specific time intervals are determined by a “Rate”, it tells you how often per seconds something happens. The unit is Hertz [Hz] defined as “how often per seconds” or “1/s”. A Sample Rate of 48kHz means that the converter takes 48,000 Samples per second.
Step Two: Quantize (or digitize)
All these Samples are still analog, for example, 1.6Volt, -0.3Volt, etc. But this analog value now has to be converted into a digital form of 1s and 0s.This is done similar to quantizing a note in GarageBand. The value (i.e. the note) cannot have any position, it has to be placed on a grid with specific values (i.e. 1/16 notes). The converter does a similar thing. It provides a grid of available numbers that the original measured Sample has to be rounded to (like when a note get shifted in GarageBand by the quantize command). This grid, the amount of available numbers, is called the Bit Depth. Other terms like Resolution or Sample Size are also used. A Bit Depth of 16bit allows for 65,535 possible values.
So the two parameters that describe the quality of an Digital Audio Signal are the Sample Rate (“how often”) and the Bit Depth (“how fine of a resolution”). The very simplified rule of thumb is, the higher the Sample Rate, the higher the possible frequency, and the higher the Bit Depth, the higher the possible dynamic.
Uncompressed Digital Audio vs. Compressed Digital Audio
So far I haven’t mentioned the “Bit Rate” yet. There is a simple formula that describes the Bit Rate as the product of Sampel Rate and Bit Depth: Sample Rate * Bit Depth = Bit Rate. However, Bit Depth and how it is used (and often misused and misunderstood) has to do with Compressed Digital Audio.
Compressed Digital Audio
First of all, this has nothing to do with a compressor plugin that you use in GarageBand. When talking about compressed digital audio, we talk about data compression. This is a special form how to encode data to make the size of the data set smaller. This is the fascinating field of “perceptual coding” that uses psychoacoustic models to achieve that data compression. Some smart scientists found out that you can throw away some data in a digital audio signal and you wouldn’t even notice it, the audio would still sound the same (or almost the same). This is similar to a movie set. If you shoot a scene on a street, then you only need the facade of the buildings and not necessary the whole building.
Although the Sample Rate is also a parameter of uncompressed digital audio, the Bit Depth is not. Instead, here is the Bit Rate used. The Bit Rate tells the encoder the maximum amount of bits it can produce per second. This determines how much data it has to throw away in order to stay inside that limit. An mp3 file (which is a compressed audio format) with a Bit Rate of 128kbit/s delivers a decent audio quality. Raising the Bit Rate to 256bit/s would increase the sound quality. AAC (which is technically an mp4 format) uses a better encoding algorithm. If this encoder is set to 128kbit/s, it produces a better audio quality because it is smarter to know which bits to throw away and which one to keep.
Conclusion
Whenever you are dealing with uncompressed audio (aiff, wav), the two quality parameters are Sample Rate [kHz] and Bit Depth [bit] (aka Resolution, aka Bit Size)
Whenever you are dealing with compressed audio (mp3, AAC), the two quality parameters are Sample Rate [kHz] and Bit Rate [kbit/s]
If you look at the Export Dialog Window in GarageBand, you can see that the Quality popup menu is different for mp3/AAC and AIFF. Hopefully you will now understand why.
Hope that helps
Edgar Rothermich
http://DingDingMusic.com/Manuals/
'I may receive some form of compensation, financial or otherwise, from my recommendation or link.'

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Since manufactured CDs are 16-bit, 44.1 KHz, is it better to record tracks in that format? It looks like STP defaults to 24-bit. Is it better to use the higher sample rate as well? 48KHz?, or should it be even higher than that?

Hi Bob. Let me see if I can remember what I learned way back in my college physics class. Keep in mind there are almost as many ill-informed opinions as there are people who record digital audio. (Oh! Dig!)
Some of that stuff sounds silly to me. Generally it's wise to use the best quality that's available to you unless you have space or bandwidth constraints. Saying you should do all your recording at 16/44.1 because it's ultimately going to end up on a CD is like saying you should shoot all your digital camera pictures at 320x200 because it's ultimately going to end up on a web page. Shooting pictures at higher resolution allows you more flexibility when editing them down for the web site, right? And audio is no different.
Here's some basics, explained in pictures:
http://www.musiciansfriend.com/document?doc_id=88273&g=home&src=3SOSWXXA
You can find more stuff like that (and endless hand waving and opinons) if you google "Bit Depth Sample Rate Physics".
Bit Depth:
Imagine in some simple world your recording software could sample waves that had amplitudes between -1 and 1. With 16 bit samples, you can record 65,536 discrete levels. (You might define a sample value of 0 to be -1 and 65,535 to be 1.) At 24 bit, you can record 16,777,216 discrete levels. The resulting representation of the wave you record will be far more accurate.
Sample Rate:
This comes down to Nyquist Frequency. The idea is that you can only record frequencies up to exactly half of your sample rate. That doesn't mean you can get a good or accurate recording of those highest frequencies, though. The higher your sample rate, the more accurate picture you can get of those higher frequencies.
This is important for recording and playback, but it's even MORE important when you later go to combine this recording with other recordings. Mixing signals, the mathematical results of processing with effects, etc. All those things will give you better results if you give them higher resolution going in.
Combining 16 low resolution tracks will give you a much worse result that combining 16 high resolution tracks and then down-converting the result. The combination of the 16 high resolution tracks will be a much more accurate representation, right?
That said, there are certainly diminishing returns. 24 bit/96kHz can give you great results, but will take up more disk space and processing bandwidth than 24/48. If you're not using superb mics and preamps, the improvement might not justify the difference. You might consider trying some experiments to see if you can detect differences yourself.

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audio sample rate
and when I check the Item Props on said clip in Browser it says the same info
I've toggled both Mono and Stereo on Capture Settings, but after capturing,
in all cases, I get the same warning message:
Audio Sample rate one your captured media files doesn't match the sample rate on source tape. This may cause it to go out of sync. Make sure the capture preset matches sample rate of tape.
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I am capturing with a Sony MiniDV camera w/ a Firewire400 cable,
and the tape is a Sony MiniDV (DVM60) Premium tape
I am unsure of the camera that shot the footage, which is from 1998
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Maximum audio sample rate and bit depth question

Anyone worked out what the maximum sample rates and bit depths AppleTV can output are?
I'm digitising some old LPs and while I suspect I can get away with 48kHz sample rate and 16 bit depth, I'm not sure about 96kHz sample rate or 24bit resolution.
If I import recordings as AIFFs or WAVs to iTunes it shows the recording parameters in iTunes, but my old Yamaha processor which accepts PCM doesn't show the source data values, though I know it can handle 96kHz 24bit from DVD audio.
It takes no more time recording at any available sample rates or bit depths, so I might as well maximise an album's recording quality for archiving to DVD/posterity as I only want to do each LP once!
If AppleTV downsamples however there wouldn't be much point streaming higher rates.
I wonder how many people out there stream uncompressed audio to AppleTV? With external drives which will hold several hundred uncompressed CD albums is there any good reason not to these days when you are playing back via your hi-fi? (I confess most of my music is in MP3 format just because i haven't got round to ripping again uncompressed for AppleTV).
No doubt there'll be a deluge of comments saying that recording LPs at high quality settings is a waste of time, but some of us still prefer the sound of vinyl over CD...
AC

I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
AC

Set sampling rate and bits on Audigy 2 Va

Hi guys, I really need your help:
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Please help me
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Can you check the rear output on the soundcard with a set of headphones, to check if that works? As long as the speaker selection is set correctly (5.), then it should work.
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A400 doesn't have anything to do with a version number. It's related to the ressource allocated to your card.
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Hi forum,
I am working on project at 44.1K, 24 bit. Audio elements are being sent to me to be added. Some have come in incorrectly; at 48K, 16 bit. I can convert easily but didn't think I needed to.
I thought by selecting the "Convert Audio Sample Rate When Importing" option when creating the project that would all be worked out.
That is what I've done - and the file seems to be the correct pitch -- yet shows up in the audio window with it's original specs (48/16). Also ... will logic keep it at 16 bit and play all other files at 24 bit?
I want to be sure about this ... something seems fishy.
Cheers
Dee, Ottawa

Hi,
I am re posting.
Regarding the same project: must everything in the arrange window of a project be the same sample rate (and bit depth). My understanding is that there is real- time conversion during playback. That all file types supported by logic - and all virtual instrument samples are converted in real time to conform to the selected bit depth and sample rate of project.
I ask only as I recieved reference sound files to temporarily place in a mix to see how mix will sit when going to Post. Two audio files are almost a semi tone higher than they are supposed to be (which is odd - so I am pretty sure it was just quickly sung in the wrong key at there end). And one file which was supposed to be timed out is not lining up.
I can work around this on this project. And I can simply convert in the sample editor and re import to compare.
But again ... I just want to check my understanding for future reference. The manual indicates differing rates etc should not be a problem. (ie: That Logic allows one to have differing rates and bit depths. Inversely, the M Sitter video implies just the opposite.
I just want to be sure of this for future reference. Any adive?
Thanks in advance.
Cheers
Dee

Audigy 4 pro vs. cubase sx sample rate - fight to the de

helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc.
I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.
in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!

Doogs wrote:
helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc. I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!
If you RTFM, you'll find out, Audigy 4 is locked into 6-bit/48kHz and 24-bit/96kHz resolutions when ASIO driver is in use.
I suppose, you still can (if not saved @ 48kHz) load your projects into SX @ 44. kHz, by selecting MME drivers instead of any ASIO.
Perhaps installing Asio4All gets it popped into that list you have there.
There are also tools to convert from 48-->44., but the source has to be as wave format.
Here is one freeware SRC tool @ http://www.voxengo.com/product/r8brain/.
.jtp

Supported bit depth and sampling frequency

Hi,
Is it possible to determine at runtime what sampling frequencies and bit depths are available for audio capture -- other than by trial and error? It does not seem that way from the API, but maybe I missed something, so I decided to post a question here. The only thing I was able to find so far is that 8-bit PCM WAV is required to be supported by all devices that support audio at all.
Thanks in advance for any info!
Sergei

Thanks for the info. I guess I was looking for something more specific - the exact bitrates and sample rates that Creative claims to support. Would you know where official and comprehensi've data can be had? There must be a tech spec somewhere.
It is common these days in business to see a recording of, say, a conference call or seminar presentation at 32k bitrate/025Hz, or even 24k bitrate/8000Hz, posted to a company's website for download by those who could not be there, and MP3 players are increasingly used for their replay. Companies use low digitization rates because there is no need for hifi and the files are much smaller: less storage, faster download.
I'd be surprized to think that Creative don't have compatibility with the standard range of rates offered by ubiquitous programs like Audacity and dBpower, the latter being one they themselves recommend!

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit. Then there is a box for sample rate, with all the different values. But it won't allow me to change it from 44100.

JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
about dithering from 32 bit to 16 bit (which is better then dithering
from 24-bit isn't it)?
I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

Creative Audigy 2 NX Bit Depth / Sample Rate Prob

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