Bit Depth & Sample Rate: 24 bit 96kHz? 192kHz?

I am using the Apogee Duet for Mac and iOS on my Mac and I love it - I'm thinking about getting an iPad for mobile recording (voice overs, mostly) and I wonder if Garage Band can manage 24 bit audio at 96 kHz or 192 kHz? I know that the Auria app can, so if nothing else I can just buy that, but since all I would use the iPad for is Voice Overs to edit later in a computer, a $50 app feels like overkill. Comments? Thoughts? Specs?

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

Similar Messages

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

Set Bit Depth to 1 bit in PS CS2 V9

I need to create documents for a scanning system. The recognition software requires the templates be Black & White, 150 dpi, at 1 bit depth. Can one set the bit depth to 1 bit in PS CS2 v9? The original document was created in MS Word, printed to Acrobat Professional 9 and converted to a TIF by AcroPro as well as acquired directly from the flat bed scanner at 150dpi in black & white mode.
Für Elise – Support the Heart & Breast Cancer Foundations

Implicit in Mr. Barnett's post is that "bitmap" in Photoshop means 1 bit per channel, with only one channel. That's what you are looking for.

Importing audio - sample rate/bit depth

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

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
rob

Filter / Sort by Bit Depth? (16 bit vs 8 bit)

I am looking for a way to use the Filter panel to sort/filter results by bit depth (16 bit vs 8 bit images) - is there a way to do this Bridge CS4? I have looked in every part of the sorting mechnisms and can't find a way. Am I missing something?

Unfortunately there is no such way in the filter panel, it might be a good
feature request, maybe you can add that using the link at the main page on
this forum.
In the meantime there are 2 possible workarounds. Regarding the fact that a
16 bit is twice the size of an 8 bit you might want to use the sort by size
criterion in the Path Bar but when having different source files or layered
files that might not always be sufficient.
If you use the find command (cmd+F / or menu Edit-Find) you can select the
find by bit depth option. Also select the option to equal and use either the
number 8 or 16.
I am looking for a way to use the Filter panel to sort/filter results by bit
depth (16 bit vs 8 bit images) - is there a way to do this Bridge CS4? I have
looked in every part of the sorting mechnisms and can't find a way. Am I
missing something?

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.'

44.1 versus 48KHz sample rates

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.

How to create a wav file from 24bit 96Khz sampling rate data

Hi
I am trying to make an VI which will play sound while acquiring data from PXI 4472 DAQ card.
My sampling rate is 96Khz and PXI 4472 card is a 24bit card.
Wave files are in 8 or 16 bit and the sampling rate is 8000, 11025, 22050 and 44100. How will I be able to play the data which I am acquiring.
How would i normalize the data into the required format needed for most of the sound cards to play.
Or are there any codec available in Windows XP which i call to play a 96KHz 24 bit sample
Does anybody ever encountered this type of problem.
Thanks in advance
Nitin

Whilst the 'standard' RIFF format specification usually accomodates 16 bit data, there is of course no reason that you can not create your own extension. It just won't be playable by Media player using the 'standard' installed drivers or codec. This may not be a problem....
WAV files can and do support other formats, you just need to know how to handle them......
There is howerver a 4GB limit (related to the pointer size in the WAV specification) which with higher bit depths on the sampling does start to become a bit of a problem.
To give you a few samples of other types of wav files check out the following site here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
The following definitions for WAV audio formats may also be of interest here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
Good luck with 24 bits.

Benefits of 96khz vs 44.1khz Sample Rate

Who knows if there are benefits to using a sample rate of 96khz instead of 44.1khz?
If so, what are they?
If not, why aren't there?

The main advantage is it lets you record higher frequencies. Whether anyone can hear those frequencies is debatable.
Other advantages are lower latency at the same buffer size (although you need more CPU power), and it's useful if you plan to do extreme pitch shifting, including using a recording as a sample in EXS. It also can make it easier for an A/D converter to do filtering properly but most modern converters can do it just fine at 44.1. For instruments that don't have higher frequencies, you won't hear a difference - it's not really more "precise". You are capturing more data, but if there are no frequencies over 20k present, that extra data is redundant and doesn't capture the waveform any more accurately than 44.1.
There's certainly benefit to going from 16 to 24 bit, but in most cases I wouldn't recommend bothering with 96 or 192.

How is actual bit depth measured

I am analyzing some recordings I made in 24-bit format in Audacity. Audacity can record true 24-bit integer files which Audition 3.0.1 recognizes as such.
After checking a couple of the files in Audition 3.0.1, I found that the meaning of "Actual bit depth" in the amplitude statistics is not entirely clear. It does seem to be based on the maximum peak in the file, but the bit estimate does not seem too clear.
For example, in one of the files if I select any portion that includes the highest peak and get amplitude statistics, the actual bit depth reported is 24. Example of a short selection that includes the peak:
Mono
Min Sample Value:    -22003
Max Sample Value:    26329
Peak Amplitude:    -1.9 dB
Possibly Clipped:    0
DC Offset:    -.003
Minimum RMS Power:    -44.45 dB
Maximum RMS Power:    -17.58 dB
Average RMS Power:    -30.18 dB
Total RMS Power:    -25.98 dB
Actual Bit Depth:    24 Bits
Using RMS Window of 50 ms
However, as far as I can tell, any selection in the same file that does not inlude the highest peak (but may include nearby close peaks) results in actual bit depth of 16:
Mono
Min Sample Value:    -20082
Max Sample Value:    22172
Peak Amplitude:    -3.39 dB
Possibly Clipped:    0
DC Offset:    -.001
Minimum RMS Power:    -54.14 dB
Maximum RMS Power:    -19.96 dB
Average RMS Power:    -36.26 dB
Total RMS Power:    -32.95 dB
Actual Bit Depth:    16 Bits
Using RMS Window of 50 ms
So it is unclear what level of peak amplitude distinguishes between 24- and 16-bit actual depth. If the bit-depth analysis is based on most-significant bits being zero, I would think that the trigger for identifying 16-bit actual depth in a 24-bit file would be to find that the 8 most-significant bits of the 24-bit samples are zero for all samples in the selection. So for a 24-bit integer file to have actual bit depth of 16 bits for a slection, the greatest peak would be less than -48 dBFS. But in the example above, the distinction seems to be having a peak amplitidue around -3.4 dB versus -1.9 dB.

>what actual difference does it make to anything?
Hard to say what difference it makes to anything without knowing what "actual bit depth" actually measures. It could be important, or could be useless. In the past I have not paid much attention to it because it is poorly described. It recently came to my attention because the files from a recent recording in 24-bit integer format were all reported as 16-bit "actual" bit depth. This is in contrast to some previous recordings made in the same way which were identified as 24-bit "actual". This implies there might be something different in the data formatting, the communication between the software and driver, between the driver and card, or something else.
It is a bit surprising that no one got Synt. to explain it properly.
>Oh, and the other thing about 24-bit int files is that they can lead you into a very false sense of security. If you decided, for instance, to reduce the amplitude of one by 48dB, then save it, and then decide to increase it again by that 48dB, you'd end up with a 24-bit file with just 16 bits of resolution - simply because it's an integer file. If you did the same thing with Audition's 32-bit floating point system, you'd lose no resolution at all.
In my workflow that produces original recordings in a 24-bit integer file format, the format is an efficient way of storing 24-bit integer data from a 24-bit card. Processing is another matter. I use the Audition preference to convert files automatically to 32-bit when opening.

Maximum bit depth-maximum render quality when dynamic linking

Hi
A bit confused by the use of Maximum bit depth and Maximum render quality as used both in Sequence Settings and also as options when rendering in AME.
1 Do you need to explicitly enable these switches in the sequence for best quality or, do you simply need to switch them on in AME when you render in Media Encoder?
2 When dynamic linking to After Effects, when should you use an 8 bit vs 16 or 32 bit working space, and, how does this bit depth interact with the maximum bit depth, maximum render quality in PPro?

Hi jbach2,
I understand your confusion. I'm like that most of the time I'm working. *chuckle* The two settings you mentioned are two completely different parameters affecting (or is it effecting) your video. You do not need to enable them within the sequence itself unless you want to preview video on you program monitor at the highest quality. I personally don't recommend it, as it's a tremendous resource hog, (the program even warns you when you try to click them) and unessecary for improving final output. Again, do not enable these options in your sequence settings if you are only wanting a high quality export. Doing so will greatly reduce your editing performance unless you have a high-end system. ...and even then I don't think its worth it unless you're editing on a huge screen with a Director who wants to see everything at a maximum quality during the edit process.
Keeping it simple...
Resizing your final output video? Use Maximum bit depth.
Starting or working with high bitdepth sources? Use Max Bit Depth.
When/where do I enable these? In the AME only. ^_^
Why?:
Enabling the Max bit and Max render only needs to be done when you are exporting. They both serve different functions.
Max Render aids in the scaling/conversion process only. My understanding is that you never need to enable the Max Render Quality (MRQ) unless you are exporting in a format/pixel ratio different from your original video. For example, when rendering a 1080p timeline out to a 480p file format, you'll want to use MRQ to ensure the best scaling with the least amount of artifacts and aliasing. If you're exporting at the same size you're working with, DON'T enable MRQ. It will just cost you time and CPU. Its only function is to do a high quality resizing of your work.
Maximum bit depth increases the color depth that your video is working with and rendering to. If you're working with video that has low color depth, then I don't believe it will matter. However, if you're working with 32 bit color on your timeline in PPro and/or After Effects, using lots of graphics, high contrast values, or color gradients, you may want to enable this option. It ultimately depends on the color depth of your source material.
The same applies to After Effects.
Create something in AE like a nice color gradient. Now switch the same project between 8,16,32 bit depth, and you will see a noticable difference in how the bit depth effects your colors and the smoothness of the gradient.
Bit depth effects how different plugins/effects change your overall image. Higher depth means more colors to work with (and incidentally, more cpu you need)
Just remember that "DEPTH" determines how many colors you can "fill your bucket with" and "QUALITY" is just that, the quality of your "resize".
http://blogs.adobe.com/VideoRoad/2010/06/understanding_color_processing.html
Check out this adobe blog for more info on color depth ^_^ Hope that helps!
----a lil excerpt from the blog i linked to above---
Now, 8-bit, 10-bit, and 12-bit color are the industry standards for recording color in a device. The vast majority of cameras use 8-bits for color. If your camera doesn’t mention the color bit depth, it’s using 8-bits per channel. Higher-end cameras use 10-bit, and they make a big deal about using “10-bit precision” in their literature. Only a select few cameras use 12-bits, like the digital cinema camera, the RED ONE.
Software like After Effects and Premiere Pro processes color images using color precision of 8-bits, 16-bits, and a special color bit depth called 32-bit floating point. You’ve probably seen these color modes in After Effects, and you’ve seen the new “32″ icons on some of the effects in Premiere Pro CS5.
jbach2 wrote:
Hi
A bit confused by the use of Maximum bit depth and Maximum render quality as used both in Sequence Settings and also as options when rendering in AME.
1 Do you need to explicitly enable these switches in the sequence for best quality or, do you simply need to switch them on in AME when you render in Media Encoder?
2 When dynamic linking to After Effects, when should you use an 8 bit vs 16 or 32 bit working space, and, how does this bit depth interact with the maximum bit depth, maximum render quality in PPro?
Message was edited by: SnJK

Photoshop CS4 - Converting bit depth on import

Photoshop seems to be converting pictures that I open to 8-bit from 24-bit: before the problem I think it was converting to 16-bit because Photoshop doesnt seem to support 24-bit but I cant remember.
I've been using Photoshop CS4 for about 4-5 weeks now without any problems and this just seems to have come out nowhere. The problem doesn't seem to be related to Camera Raw at all because I don't use it to import my pictures, so the "Workflow Options" don't seem to have any effect on the converting. I've also tried to delete the preferences file and that doesn't help either.
Any ideas?

Photoshop doesn't convert bit depths or color modes when opening documents, unless there is an unsupported bit depth (like 12 bits/channel) in which case we convert upward (to 16 bits/channel).
It would help if you told us what file format you were talking about, and what you mean by "24 bit".

Showing file attributes/metadata (e.g. sampling rate) in Finder

Hi, I am a new MacBook Pro user. My first computer was an Apple II, but I have spend my past two decades using PC. I recently bought a MacBook Pro and am very happy with it. I have used it to write music (Sibelius), process recordings (Sound Studio/GarageBand), burn my own CDs, create slideshows (iPhoto) and a DVD (iDVD) in this past month.
However, I find the general 'information hiding' attitude really annoying. The Mac Finder *****. I have thousands of wave files and mp3 files originally created on PCs and copied over to Mac. I need to be able to browse these in hundreds of folders. But The Finder will not show me file attributes other than name, data, size, kind and comment. I need to see other audio attributes such as sampling rate, bit depth, etc.
I do not want to use iTune. It is not a file/folder browser. I know where my files are in the folder hierarchy. I don't need iTune to help me using its flat 2D table view. Besides, I also have thousands of files of other types (e.g. jp3, png, etc). I am not sure iTune can help me look at attributes of these files.
How in world does someone get this type of view on Mac? I have looked at Xfolders, File Sheriff, A Better Finder Attributes and DiskLibrary. None of these provide what I want. I found RAGE Mac Explorer to be extremely attractive. it's got a lot of browsing features. But it still doesn't want to show me attributes other than those shown by the Finder.
By browsing this forum, I realized that I missed another possible candidate: PathFinder. I will try this next. I see posts from 2005 talking about the same problem and proposing PathFinder as the only solution. I simply can't believe such simple tasks cannot be easily done, out of box, without needing to pay for third-party software (if I can find one, that is).
Help!
Fred

This is becoming a monologue. I gave PathFinder a try. I like it a lot. In fact, it is even more powerful in terms of file browsing compared to RAGE Mac Explorer. In particular I get much more detailed "GetInfo" info from PathFinder. It basically returns metadata similar to Preview Info. And you can get these (e.g. sampling rate) from: 1) last Preview column in Column view, 2) GetInfo, 3) Inspector-like permanent preview pane at the bottom of List view, and 4) File Report. You can also show a short "preview" string for each file in List view (similar to the string shown in Finder's icon view).
But... I still can't find a way to add metadata I care about to columns in PathFinder, the way you can in Windows Explorer.
So, I figure my last resort is to create some folder automation to gather info about each file according to its file type, and modify Spotlight Comment for each file. At least in the Finder I can show the Comment field. But oddly, neither PathFinder nor RAGE Mac Explorer lets me show the Comment field.
Sigh.
Fred

Problems with sample rate

hello:
I have a sound blaster audigy platinum ex and I'm trying to make some recordings with adobe audition, but this program works at 44khz and the sound card seems to work only at 48khz, so it doesn't let me recording. how can I change the sampleing rate of the card?
thanks.

By setting it in Audition audio settings. W/ ASIO device drivers you can have 16-bit/48kHz (and IIRC, 24-bit/96kHz) mode(s) only). For other modes you need to use other device driver mode. Also, Asio4All should handle all modes (16/24-bit and 44.1, 48 and 96kHz).
- http://www.asio4all.com
The quality in 44.1kHz mode is POOR for your card. If you switch to 24-bit/44.1kHz quality is otherwise better but the frequency gets cutted already around 15kHz.
You get the best quality by setting the sample rate to 96kHz (16 and 24 bit resolution are ~same in these cards) for recordings (for project?) and then when export the final mix just use SRC to CD quality (Audition do have quite good SRC quality but, if you use additional software for SRC (like Voxengo r8brain (free)), export the final mix using 96kHz and then execute SRC).
Something more on Creative ASIO -
http://forums.creative.com/creativel...essage.id=1726
jutapa
Message Edited by jutapa on 11-09-200602:43 PM

Bit Depth & Sample Rate: 24 bit 96kHz? 192kHz?

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